| 00:03.27 | *** join/#asterisk BelTechs (~BelTechs@72.87.189.51) |
| 00:06.49 | BelTechs | Hello, Im trying to debug my trunk. The settings that I am using are known working setting on a *1.6 I have tried 3 different *1.8 distros with the same settings and get the same results. sip debug http://pastebin.com/DikQb0hw |
| 00:06.55 | BelTechs | Thanks in advance |
| 00:18.45 | BelTechs | Hello, Im trying to debug my trunk. The settings that I am using are known working setting on a *1.6 I have tried 3 different *1.8 distros with the same settings and get the same results. sip debug http://pastebin.com/DikQb0hw Thanks in advance |
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| 02:17.23 | kaldemar | BelTechs: what are the results? your pastebin shows successful registrations. |
| 02:18.07 | yruss | hi |
| 02:28.31 | yruss | Pls, some help for a telephone newbie. Can smbd explain me about fundamentals? Let's imagine - I have an goautodialnow running on the server with asterisk - it connected to my internet router - i guess it needs an outgoing telephone line to make calls? So the question is can it be an some fxo\fxs adapter with external tel. line connected thru lan to my router? Or is should be only some pci card with fxo\fxs on the server? |
| 02:30.02 | kaldemar | it can be both. or a gsm gateyway, pri/bri hardware or you can use an ITSP. |
| 02:30.15 | kaldemar | ~book |
| 02:30.16 | infobot | For more information about the Asterisk book, see ~thebook |
| 02:30.23 | kaldemar | ~thebook |
| 02:30.24 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 02:30.59 | yruss | ok thanx) |
| 02:41.02 | *** join/#asterisk cj (~cjac@adsl-207-32-169-17.rockisland.net) |
| 02:41.05 | cj | hey all |
| 02:41.16 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 02:41.39 | *** join/#asterisk luisfelice (~luisfelic@190.39.198.95) |
| 02:41.48 | cj | I'm calling in to my asterisk machine and I get the following in the console: |
| 02:41.54 | cj | [Apr 20 19:41:15] WARNING[30164]: chan_dahdi.c:7608 handle_alarms: Detected alarm on channel 4: Red Alarm |
| 02:41.57 | cj | [Apr 20 19:41:16] NOTICE[30136]: sig_analog.c:3680 analog_handle_init_event: Alarm cleared on channel 4 |
| 02:42.00 | luisfelice | Hello all, how secure is the md5 authentication? |
| 02:42.19 | cj | luisfelice: go with sha-256 :) |
| 02:43.26 | luisfelice | cj: my hardphone only supports md5, but I would like to know if it is secure to authenticate with md5 through internet |
| 02:43.55 | luisfelice | cj: using IAX2 |
| 02:44.07 | cj | http://www.nsrl.nist.gov/collision.html |
| 02:44.23 | cj | luisfelice: it's relatively secure |
| 02:44.36 | cj | but you can't use it for US government contracts at this point, I hear |
| 02:45.01 | luisfelice | cj: thanks |
| 02:45.10 | cj | sure :) |
| 02:46.02 | cj | so, I've set the default context in chan_dahdi.conf to incoming, and I've added the following to my extensions.conf: |
| 02:46.05 | cj | [incoming] |
| 02:46.08 | cj | exten => s,1,Answer() same => n,Playback(tt-weasels) same => n,Hangup() |
| 02:46.23 | cj | irssi gobbled the newlines, of course |
| 02:46.56 | cj | I expect that calling my PSTN number should result in hearing the weasels audio |
| 02:47.14 | cj | but all I get is the above WARNING / NOTICE messages |
| 02:48.11 | cj | what might I be doing wrong? |
| 02:50.00 | cj | tries fxsls instead of fxsks |
| 02:54.01 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
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| 03:06.58 | cj | nah, that didn't work |
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| 04:11.08 | kuku | Anyone used google analytics for call tracking ? Wondering how to get started on that. |
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| 04:28.52 | OldMonk | hi |
| 04:29.48 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 04:31.15 | OldMonk | weird problem with connecting to asterisk from ekiga on the local server: http://pastebin.ca/2048866 |
| 04:32.20 | OldMonk | sip.conf: http://pastebin.ca/2048867 |
| 04:32.41 | OldMonk | i added the port 5061 to test whether that was the problem, but no joy |
| 04:32.42 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 04:41.33 | OldMonk | well, the same thing works with SFLPhone, so i guess i'll just use that for testing |
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| 05:54.50 | *** join/#asterisk kwk (~kleine@212.8.6.6) |
| 05:55.27 | kwk | Hi there |
| 06:02.18 | kwk | Yesterday I've opened an issue ticket (https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=19154) and it already has been closed with this comment: "Specifically, you are not using your peer definitions for your research peers." Can anybody please have a look at my issue and maybe explain what might be meant by this problem? |
| 06:03.20 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
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| 06:20.20 | kaldemar | kwk: replace Dial(SIP/1690:1234::research2@192.168.51.253,15) with Dial(SIP/research1/1690) |
| 06:21.19 | *** join/#asterisk jkroon (~jkroon@dsl-241-228-173.telkomadsl.co.za) |
| 06:24.39 | kaldemar | kwk: and fromuser needs some thinking, when you dial research2 from research1 with SIP/research2/<number>, you should have fromuser=research1 under [research2] in research1 sip.conf. as the sample config states, fromuser is "how your provider knows you". |
| 06:30.07 | kaldemar | kwk: UPGRADE-1.6.txt will also tell you that the username option was renamed to defaultuser. despite it says that the username option will be removed in the next version, it still seems to be supported in 1.8.3.2 code but with a note of deprecation. |
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| 07:49.29 | kwk | kaldemar: Could you please show me how you'd write your configs. Here's my basis: http://www.pasteall.org/21009/ini |
| 07:50.33 | *** join/#asterisk daxt (~daxt@112.135.84.148) |
| 07:51.02 | daxt | guys , i am getting an error in my CentOS system when i try to install asterisk 1.6 , this is the error http://pastebin.com/hKsEpVxG , how do i resolve this ? |
| 07:51.08 | kaldemar | kwk: i already told you how to start. |
| 07:52.05 | wdoekes2 | daxt: uninstall asterisk 1.4? |
| 07:52.38 | daxt | wdoekes2 hi , yes i did and deleted all folders which asterisk were on |
| 07:53.04 | wdoekes2 | no you didn't |
| 07:53.27 | daxt | :O |
| 07:53.30 | *** join/#asterisk Tim_Toady (~moi@62.1.175.227) |
| 07:54.03 | daxt | what folders i should look for if i am to properly uninstall asterisk ? |
| 07:54.17 | wdoekes2 | I doubt that your package manager is lying: you'll need to resolve your package manager issues first |
| 07:54.37 | wdoekes2 | daxt: if you're using packages, you should NOT be touching folders.. the package manager should take care of that |
| 07:55.25 | daxt | I doubt that your package manager is lying: you'll need to resolve your package manager issues first <--- give me a clue please |
| 07:55.43 | wdoekes2 | I don't use centos.. on debian I'd do: apt-get -f install |
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| 07:57.02 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 07:57.06 | daxt | wdoekes2 ok i'll ask from CentOS room about it :) thanks |
| 07:57.40 | kaldemar | daxt: you should use yum to remove packages if you used it to install them, not delete any folders by hand. |
| 07:58.11 | kaldemar | "yum remove asterisk14" |
| 07:59.22 | daxt | i did that and removed folders where Asterisk usually lives |
| 08:04.36 | RypPn | rpm -qa | grep asterisk |
| 08:05.51 | kwk | kaldemar: thank you for your help. Here's how I've adopted your suggestions http://www.pasteall.org/21010 (side-by-side view) Is this what you've meant? Please note, that these changes didn't work. |
| 08:07.50 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 08:07.54 | kaldemar | kwk: how do they not work? pastebin CLI output with sip debug for a call. |
| 08:09.17 | kwk | kaldemar: ok I will do so, but first I'll have to log out and reconnect on the machine from where i can administer the asterisk VM. will be back in a few minutes. |
| 08:09.49 | *** join/#asterisk kwk (~kwk@carbon.gonicus.de) |
| 08:10.25 | jacc0 | I'm makeing an asterisk fail-over server and I'm using asterisk 1.8 realtime; we are doing mysql replication betweeen servers -> how can I transport active sip registrations? |
| 08:10.55 | jacc0 | do I have to change realtime chash settings or something? |
| 08:14.06 | *** join/#asterisk daxt (~daxt@112.135.95.62) |
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| 08:20.46 | jacc0 | kaldemar: any usefull tips for me |
| 08:20.55 | jacc0 | I'm using flip1405 |
| 08:24.22 | daxt | kaldemar r u around ? i resolved that issue and ran rum remove asterisk and rebooted the server , but i still can connect to asterisk server with asterisk -r , why asterisk is still there even after removing the package ? |
| 08:26.54 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 08:36.30 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 08:36.43 | kwk | kaldemar: Here is my CLI output: http://www.pasteall.org/21014 |
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| 08:48.02 | kaldemar | kwk: which machine was that? |
| 08:48.16 | kwk | research1 |
| 08:48.28 | kwk | kaldemar: research1 |
| 08:49.26 | kaldemar | "Unable to create channel of type 'SIP'" looks really strange. |
| 08:49.44 | kwk | kaldemar: do research1 needs to be able to ping research2? |
| 08:49.56 | kwk | s/do/does |
| 08:50.03 | kaldemar | ah, they haven't registered to each other. |
| 08:50.47 | kaldemar | make research1 and research2 register to each other in sip.conf. in research1 you put register => research1:1234@ip_of_research2 and vice versa. |
| 08:51.37 | *** join/#asterisk jkroon (~jkroon@dsl-240-159-23.telkomadsl.co.za) |
| 08:51.44 | kwk | kaldemar: ok, will do. mom. pls |
| 08:51.46 | kaldemar | not ping necessarily but they should be able to communicate in UDP via the used SIP port. |
| 08:54.45 | kwk | kaldemar: here's the CLI output without SIP debug (next one will have it again): http://www.pasteall.org/21015 |
| 08:56.37 | kaldemar | kwk: see what happens on research2 CLI when you make the call. |
| 08:57.03 | kwk | kaldemar: sip debug for research1: http://www.pasteall.org/21016 |
| 08:57.52 | kwk | kaldemar: sip debug for research2: http://www.pasteall.org/21017 |
| 08:58.27 | kaldemar | kwk: your peer definitions lack a context. that's probably causing the forbidden. with your current dialplan, put context=default under [research1] and [research2]. |
| 09:00.49 | kwk | kaldemar: not working: http://www.pasteall.org/21018 |
| 09:01.24 | kwk | kaldemar: what's with the message from research2? "username mismatch, have <research1>, digest has <research2>" |
| 09:02.12 | *** join/#asterisk schmidts (~schmidts@2001:858:4:0:66b9:e8ff:fec3:ae88) |
| 09:02.14 | schmidts | hello |
| 09:04.35 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 09:06.34 | kaldemar | kwk: remove the defaultuser option |
| 09:09.15 | kaldemar | kwk: and put usernames in the registration statements, register => research1:1234@ip2/research2 and register => research2:1234@ip1/research1 |
| 09:10.42 | *** join/#asterisk wonderworld (~ww@port-92-201-26-19.dynamic.qsc.de) |
| 09:10.55 | kwk | kaldemar: nope, the callee responses with "username mismatch, have <research1>, digest has <research2>" |
| 09:11.57 | kaldemar | what do your register statements look like? |
| 09:12.47 | kwk | on research1: |
| 09:12.49 | kwk | register => research1:1234@192.168.51.88/research2 |
| 09:12.51 | kwk | on research2: |
| 09:12.53 | kwk | register => research2:1234@192.168.51.253/research1 |
| 09:13.55 | kaldemar | and sip show peers? |
| 09:13.59 | kwk | mom |
| 09:14.12 | kwk | research1*CLI> |
| 09:14.14 | kwk | research1*CLI> sip show peers |
| 09:14.16 | kwk | Name/username Host Dyn Forcerport ACL Port Status |
| 09:14.18 | kwk | 1690 (Unspecified) D 0 UNKNOWN |
| 09:14.20 | kwk | 1691/1691 192.168.51.238 D 5060 OK (9 ms) |
| 09:14.22 | kwk | research2/research2 192.168.51.88 D 5060 Unmonitored |
| 09:14.24 | kwk | 3 sip peers [Monitored: 1 online, 1 offline Unmonitored: 1 online, 0 offline] |
| 09:14.26 | kwk | research1*CLI> |
| 09:14.28 | kwk | research2*CLI> sip show peers |
| 09:14.30 | kwk | Name/username Host Dyn Forcerport ACL Port Status |
| 09:14.32 | kwk | 1689/1689 192.168.51.230 D 5060 OK (12 ms) |
| 09:14.34 | kwk | research1/research1 192.168.51.253 D 5060 Unmonitored |
| 09:14.36 | kwk | 2 sip peers [Monitored: 1 online, 0 offline Unmonitored: 1 online, 0 offline] |
| 09:14.38 | kwk | research2*CLI> |
| 09:14.57 | kaldemar | research2/research2 and research1/research1 is not correct. you should see research2/research1 and research1/research2. |
| 09:20.23 | kaldemar | if the registration approach doesn't stick, put the defaultuser options back in there. |
| 09:22.43 | kwk | kaldemar: maybe it would be easier to have two accounts for incoming and outgoing connections per sip.conf ?! |
| 09:28.16 | kwk | kaldemar: one as peer and one as friend |
| 09:28.37 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 09:28.39 | Dovid | morning all |
| 09:29.00 | angryuser | Dovid, well that depends on gmt |
| 09:29.15 | Dovid | lol |
| 09:29.23 | Dovid | when ever i come here it's morning |
| 09:29.27 | cusco_ | morning :) |
| 09:29.29 | Dovid | its actually 12:29 PM where i am |
| 09:29.38 | cusco_ | 10.29 am |
| 09:29.50 | cusco_ | gmt +0 +daylight savings |
| 09:29.59 | angryuser | Dovid, i am plus 1h from you |
| 09:30.07 | cusco_ | greet well suited to me :p |
| 09:30.55 | kaldemar | kwk: a friend is both peer and user. |
| 09:31.26 | cusco_ | kwk: sip show register |
| 09:31.29 | cusco_ | or registry |
| 09:33.47 | kwk | cusco_: http://www.pasteall.org/21024 |
| 09:34.19 | kaldemar | kwk: http://ofps.oreilly.com/titles/9780596517342/asterisk-OutsideConn.html#OutsideConnectivity_id291273 |
| 09:35.54 | kaldemar | kwk: what version of asterisk are you using btw? |
| 09:36.36 | kaldemar | it says 1.8.3 in the issue report... |
| 09:37.39 | *** join/#asterisk trintiron (~djd@89.184.141.142) |
| 09:40.43 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 09:45.54 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
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| 10:00.16 | kwk | kaldemar: 1.8.3.2 |
| 10:02.38 | *** join/#asterisk usc911 (~ben@78-105-116-233.zone3.bethere.co.uk) |
| 10:02.53 | usc911 | anyone have an issue with echo on their systems? I havean initial echo that fades out after maybe 5 seconds |
| 10:05.42 | Dovid | usc911: PSTN or SIP ? |
| 10:06.14 | usc911 | sip |
| 10:06.20 | usc911 | got 10 snom 300's |
| 10:06.46 | kwk | will be back in an hour and will try another more reasonable sip.conf with one user defined with two types: |
| 10:06.48 | kwk | [user1] |
| 10:06.50 | kwk | type=peer |
| 10:06.52 | kwk | [user1] |
| 10:06.54 | kwk | type=user |
| 10:07.42 | *** join/#asterisk micols (~ident@rlogin.dk) |
| 10:08.33 | usc911 | thanks man |
| 10:23.23 | *** join/#asterisk trintiron (~djd@89.184.141.142) |
| 10:24.42 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 10:25.04 | jkprg | hi. do you know about some app that would allow to move established SIP conversation to conference ( confbridge for instance)? thx |
| 10:27.16 | kaldemar | jkprg: app as in a dialplan application? |
| 10:27.36 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 10:27.42 | jkprg | kaldemar: yes. |
| 10:27.56 | kaldemar | used by who? a third party? |
| 10:28.41 | jkprg | kaldemar: I just want to moved conversation to conference the most friendly way.... |
| 10:29.22 | kaldemar | who is supposed to initiate it? |
| 10:30.02 | *** join/#asterisk daxt (~daxt@112.135.88.244) |
| 10:30.19 | jkprg | kaldemar: either caller or callee |
| 10:31.42 | jkprg | kaldemar: right now I'm doing xfer to conference and call conference. I'd like to do it in one step if it's possible |
| 10:32.50 | jkprg | kaldemar: moreover without interruption (or do it fast so people won't notice) :-) |
| 10:34.44 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:98ef:ec93:3018:b1b8) |
| 10:37.57 | kaldemar | jkprg: i'd look into Dial options tTFg. |
| 10:40.48 | jkprg | kaldemar: thx. maybe I've found it http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO |
| 10:43.51 | jacc0 | extracted 19 database names and 266 table names |
| 10:44.12 | jacc0 | wrong chat |
| 10:44.15 | jacc0 | ;) |
| 10:46.24 | jacc0 | don't ask |
| 10:46.26 | jacc0 | :p |
| 11:11.01 | *** join/#asterisk dimm1 (~appleworm@unaffiliated/dimm) |
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| 11:54.43 | flashdeluxe | does anybody know how i can implement my capi device in the asterisk-gui? |
| 11:57.36 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:290a:c9e6:2b0b:b) |
| 12:01.33 | *** join/#asterisk mawhii (~mawhii@170.220.119.70.cfl.res.rr.com) |
| 12:07.33 | kwk | kaldemar: I've tried the configuration from here (http://ofps.oreilly.com/titles/9780596517342/asterisk-OutsideConn.html#OutsideConnectivity_id291273) and adopted it to my IP configuration. The research1 and research2 servers can be pinged |
| 12:07.35 | kwk | now using i.e. "ping serverA". The issue remains: directmedia=yes causes this warning: "[Apr 21 14:01:02] WARNING[3431]: chan_sip.c:19184 handle_response_invite: just did sched_add waitid(2116) for sip_reinvite_retry for dialog 0490ca5b4a5c451b25b312da224b340f@192.168.51.253:5060 in handle_response_invite" |
| 12:07.37 | kwk | with directmedia=no, everything works perfectly. |
| 12:08.48 | *** join/#asterisk cerberus_za (~coert@196-210-190-65.dynamic.isadsl.co.za) |
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| 12:13.28 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 13:07.27 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 13:11.21 | jacc0 | I'm makeing an asterisk fail-over server and I'm using asterisk 1.8 realtime; we are doing mysql replication betweeen servers -> how can I transport active sip registrations? do I have to change realtime chash settings or something? |
| 13:11.49 | jacc0 | becasue devices don't work untill the re-register |
| 13:13.49 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 13:20.34 | ixyd_ | maybe i am wrong, but i think the registrations are taken from the astdb and not from the realtime?! |
| 13:25.30 | *** join/#asterisk JonnyD_work (~Jon@173.226.80.154) |
| 13:28.45 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
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| 13:44.12 | *** join/#asterisk TheCompWiz (~TheCompWi@wsip-68-109-200-102.mc.at.cox.net) |
| 13:46.34 | TheCompWiz | can someone help me figure out how to manipulate the caller-id that is used when doing "SendText" ... |
| 13:47.02 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 13:47.37 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 13:47.44 | TheCompWiz | is there some magic I can do prior to the Answer()? |
| 13:48.02 | JerJer | TheCompWiz: Set(CALLERID(num)=12345)) ? |
| 13:48.18 | TheCompWiz | I'll give it a shot JerJer. |
| 13:49.15 | JerJer | Set(CALLERID(num)=0000000000) |
| 13:50.18 | *** join/#asterisk puzzled_ (~foobar@5ED2AE82.cm-7-3c.dynamic.ziggo.nl) |
| 13:50.54 | TheCompWiz | grrr.... still has the extension I dialed's number... |
| 13:52.50 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:52.50 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:53.10 | *** join/#asterisk halindrome (~ahby@d-link.aptest.com) |
| 13:55.11 | TheCompWiz | any other suggestions on how to modify the "from" when using SendText? |
| 13:56.37 | JerJer | I had one to many ) last time (did you catch that?) |
| 13:56.41 | TheCompWiz | yeah. |
| 14:05.30 | iprouteth0 | is anyone able to currently call outbound using google voice through asterisk? |
| 14:06.37 | *** part/#asterisk usc911 (~ben@78-105-116-233.zone3.bethere.co.uk) |
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| 14:17.28 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 14:17.53 | bent_screwdriver | anyone using the latest polycom sip 3.3.1? |
| 14:18.07 | *** join/#asterisk coppice (~chatzilla@62.166.232.220.dyn.pacific.net.hk) |
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| 14:35.08 | TheCompWiz | grrrr.... flipping through source... and it doesn't look like there is a way to set the caller id.... |
| 14:36.41 | *** join/#asterisk cyford (~cyford@adsl-074-188-021-226.sip.asm.bellsouth.net) |
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| 14:46.54 | TheCompWiz | bent_screwdriver: I did use the UC firmware for a bit... and went back to the 3.2.3 firmware. the UC stuff requires brand-new configs for your phones... |
| 14:48.28 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 14:48.38 | bent_screwdriver | TheCompWiz: saw that...and ran the conversion utility which actually worked on my configs. just playing around now but wasn't sure which sip.cfg is the base config now....looks like maybe sip-basic and sip-interop? |
| 14:49.03 | malcolmd | iprouteth0: haven't tried today. more google oddness? |
| 14:49.24 | iprouteth0 | I can receive inbound, but outbound fails |
| 14:51.24 | *** part/#asterisk kwk (~kwk@carbon.gonicus.de) |
| 14:53.21 | bent_screwdriver | TheCompWiz: curious if they've now enabled direct call pickup on a ringing buddy button, or at least made the BFL blink or something.... |
| 14:53.46 | halindrome | I know there are a million factors that *might* influence this, but... I have an asterisk server that has an external IP address, and I cannot seem to have a SIP phone connect to it reliably from anywhere except on the same LAN or if I use a VPN. suggestions on where to start looking? |
| 14:54.34 | keith4 | what are the symptoms of your "unreliable" connection? |
| 14:57.14 | *** join/#asterisk clintc (~clintc@n128-227-125-7.xlate.ufl.edu) |
| 14:58.08 | iprouteth0 | halindrome: have your forwarded your sip port to asterisk server? |
| 14:58.31 | halindrome | the asterisk server has an external IP address - shouldn't require any forwarding |
| 14:58.41 | iprouteth0 | ok, quite right... |
| 14:58.49 | halindrome | keith4: I shouldn't have said unreliable. It never connects at all |
| 14:59.03 | *** join/#asterisk nick^^ (~nick@24-117-243-164.cpe.cableone.net) |
| 14:59.17 | halindrome | ... nothing in the logs on the asterisk server to indicate that a phone is trying to connect. which leads me to believe firewall. But the firewall is open from the source IP address |
| 14:59.51 | nick^^ | exit |
| 14:59.52 | iprouteth0 | I'm successfully able to do so, however my asterisk box is behind nat, port is forwarded, and I believe extension is set to use nat |
| 15:00.08 | iprouteth0 | worked successfully with andriod mobile using sip softphone |
| 15:00.08 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 15:00.13 | *** join/#asterisk philm (~root@163.205.87.85) |
| 15:00.17 | keith4 | ~sipnat |
| 15:00.17 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 15:00.53 | keith4 | halindrome: did you remember to allow UDP through the firewall? |
| 15:01.06 | halindrome | keith4: I *think* so. checking |
| 15:02.02 | halindrome | keith4: yes - the rule is 'all' |
| 15:03.06 | keith4 | regardless, if it works fine on the LAN, and via VPN... how is that *not* a network/firewall problem? ;-) |
| 15:03.30 | iprouteth0 | maybe use netcat or nmap to check if port is open and or connect to it? |
| 15:05.09 | halindrome | keith4: well - here's were it gets weird. it ALSO worked without a VPN from one remote office. In that case the remote office had a dd-wrt router, the sip phone was behind that and NAT. connected up and worked perfectly. so weird |
| 15:05.11 | iprouteth0 | damn google.... |
| 15:05.38 | halindrome | iprouteth0: you mean connect to port 5060 |
| 15:06.08 | *** join/#asterisk MrSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 15:06.12 | *** join/#asterisk nick^^ (~nick@24-117-243-164.cpe.cableone.net) |
| 15:06.22 | iprouteth0 | right. netcat might give you more useful info that the endpoint |
| 15:06.26 | iprouteth0 | just a suggestion |
| 15:07.37 | halindrome | thanks I will look at that too. also checking asterisk to see if it THINKS it is behind NAT and maybe that is confusing it |
| 15:09.18 | iprouteth0 | damn. I really wish I could make outbound through google again. I can't for the life of me figure out what happened |
| 15:10.19 | halindrome | Mine still works, but not reliably. FWIW. It seems like the jabber signaling doesn't always come through |
| 15:13.37 | iprouteth0 | i just did a jabber debug, but have no idea if I've found anything. Seems like more useful info than any other troubleshooting I've sone so far |
| 15:14.37 | iprouteth0 | here is my jabber debug if anyone can give assistance http://pastebin.com/SaxyF4wW |
| 15:15.28 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 15:16.08 | halindrome | iprouteth0 and keith4: So.... I am super stupid. the asterisk server had 'NAT' set to yes even though it is on an external IP. changed that and viola! |
| 15:16.19 | iprouteth0 | :) |
| 15:21.43 | *** join/#asterisk rubberneck (~chatzilla@ext-52.sagetelecom.net) |
| 15:23.32 | rubberneck | I am getting the following error after upgrading to 1.8 from 1.6 """ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: =1""" This is caused by a gosub that calls the following line """GoToIf($[${ARG1}=18003575534]?s,cidfix)""" Do I need quotes now or something? |
| 15:25.31 | *** join/#asterisk nick^^ (~nick@24-117-243-164.cpe.cableone.net) |
| 15:26.44 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 15:27.20 | *** join/#asterisk hetii (~hetii@87.99.51.172) |
| 15:27.25 | hetii | Hello :. |
| 15:27.27 | hetii | :) |
| 15:31.08 | rubberneck | I have never seen it this quiet in here....... |
| 15:31.44 | keith4 | shhhhhh |
| 15:33.17 | ChannelZ | Everyone's off playing Portal 2 |
| 15:33.30 | rubberneck | No doupt |
| 15:34.06 | hetii | Q: about this file: http://pastebin.com/wKRwmpP0 |
| 15:34.30 | hetii | why there are # |
| 15:34.33 | hetii | group=0,11 and group = 63 |
| 15:35.35 | hetii | what meant the number after 0, and why when i call -- DAHDI/i1/19223-33 is ringing i see that this call is made by channel 7 ? |
| 15:35.58 | hetii | <PROTECTED> |
| 15:36.12 | hetii | dahdi show channels |
| 15:36.31 | hetii | ? |
| 15:36.51 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 15:37.07 | hetii | general i had trouble to mad a call outside by my dadhi trunk the incoming works fine:) |
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| 15:43.13 | *** join/#asterisk Tim_Toady (~moi@62.1.175.227.dsl.dyn.forthnet.gr) |
| 15:44.59 | nick^^ | hetii: i think the second # after group is usually for the D channel |
| 15:47.39 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:48.40 | rubberneck | Can someone tell me why on v 1.8.3 """GoToIf($[${ARG1}=18003575534]?s,cidfix)""" errors with """syntax error, unexpected '=', expecting $end""" |
| 15:51.57 | nick^^ | rubberneck: maybe try double quote around ${ARG1} and =18003575534 |
| 15:51.58 | keith4 | that will probably fail if ARG1 is empty |
| 15:52.03 | keith4 | yah, do what nick^^ said |
| 15:54.00 | rubberneck | nick^^: Ok, testing..... |
| 15:54.04 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 15:58.34 | hetii | i got: -- Channel 0/1, span 1 got hangup request, cause 1 |
| 15:58.41 | hetii | or casuse 16 or 18 |
| 15:58.56 | hetii | is there some book where i can read what the cause code means? |
| 15:59.44 | *** join/#asterisk iratik (~Adium@74-84-99-12.client.mchsi.com) |
| 16:00.43 | TheCompWiz | hetii: http://www.voip-info.org/wiki/view/Asterisk+cmd+Hangup |
| 16:00.50 | iratik | I totally realize this probably isn't the place to ask about digium TDM cards… but On the TDM400p (Wildcard) … Can you basically configure it to be either 4 FXO or 4 FXS dynamically .. or do you have to have specific hardware to convert the ports between the types? |
| 16:01.49 | TheCompWiz | is there a way to modify the contacturi within a dialplan? |
| 16:05.28 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v001-181.mobile.uci.edu) |
| 16:06.45 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
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| 16:14.22 | *** part/#asterisk halindrome (~ahby@173-160-98-68-Minnesota.hfc.comcastbusiness.net) |
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| 16:15.26 | *** part/#asterisk [T]ank (~Tank@206.71.78.158) |
| 16:16.11 | keith4 | iratik: the FXO and FXS modules are different hardware |
| 16:16.32 | keith4 | e.g., the little green and red add-on modules |
| 16:21.04 | *** join/#asterisk nick^^ (~nick@24-117-243-164.cpe.cableone.net) |
| 16:21.36 | rubberneck | Looks like you can not pass an argument to a subroutine from a subroutine in 1.8, anyone else come across this? |
| 16:25.27 | *** part/#asterisk hetii (~hetii@87.99.51.172) |
| 16:25.31 | *** join/#asterisk nickfennell (~nick@cov1.appliansys.com) |
| 16:25.34 | nickfennell | hey peeps |
| 16:25.45 | nickfennell | is it possible to find out which context an extension exists within ? |
| 16:26.22 | keith4 | nickfennell: look at the dialplan? ;-) |
| 16:26.43 | nickfennell | hmm, i have elastix ontop |
| 16:27.02 | nickfennell | I have a call supposedly coming into default where it should contact a ring all group but I get a 404 |
| 16:27.09 | nickfennell | I assume ring all not in default :) |
| 16:28.36 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:28.45 | nickfennell | OoooooooOOOO |
| 16:28.48 | nickfennell | dialplan show! |
| 16:28.50 | nickfennell | CLI ftw! |
| 16:30.06 | nickfennell | If I'm bringing in external SIP |
| 16:30.11 | nickfennell | what's the best context? |
| 16:30.16 | nickfennell | my AsteriskFoo is aged. |
| 16:32.22 | paulc | nickfennell: make one! like [in-sip] :) |
| 16:32.41 | paulc | then set whatever it is in the [general] section of sip.conf, as context=in-sip |
| 16:32.44 | nickfennell | super lame, If I set the register string to /101 it rings, if I set it to /600 it doesn't |
| 16:32.49 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 16:39.55 | *** part/#asterisk iratik (~Adium@74-84-99-12.client.mchsi.com) |
| 16:43.04 | nickfennell | If I want to bring calls in but use the Inbound Routes feature, A) What context for Inbound calls, B) what should my register string look like? |
| 16:43.14 | nickfennell | any ideas? |
| 16:44.23 | paulc | nickfennell: what is the "Inbound Routes feature" - are you using Trixbox, Freepbx, etc? |
| 16:45.06 | nickfennell | Yeah exactly |
| 16:45.11 | nickfennell | Which is why this is so tricky |
| 16:45.20 | nickfennell | trying to figure out how they've built it |
| 16:45.28 | nickfennell | It's Elastix which is basically FreePBX |
| 16:46.36 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 16:48.52 | *** join/#asterisk ixyd_ (~denzs@ip-78-94-178-44.unitymediagroup.de) |
| 16:49.32 | *** part/#asterisk ixyd_ (~denzs@ip-78-94-178-44.unitymediagroup.de) |
| 16:50.13 | keith4 | nickfennell: anyone not using Elastix/FreePBX (read: most of this channel) will have no idea what you're talking about, myself included |
| 16:53.24 | paulc | #freepbx might be a better place to ask |
| 16:53.40 | nickfennell | Yeah valid. |
| 16:53.51 | nickfennell | the dialplan show was handy though |
| 16:53.59 | paulc | Anyone else watching the Asterisk Tech Tips webinar (in 7 minutes)? |
| 16:55.12 | malcolmd | me :D |
| 16:57.03 | russellb | leifmadsen and I are about to start a webinar. www.asterisk.org/techtips |
| 16:57.09 | russellb | in about ... 3 minutes |
| 16:57.11 | paulc | malcolmb: haha of course :) |
| 16:57.13 | therawr | WEBINAR |
| 16:58.32 | keith4 | the "system requirements" only lists Windows and Mac, so I guess I'm out |
| 16:58.58 | russellb | yeah :-( |
| 16:59.02 | russellb | we didn't pick the software |
| 17:00.34 | *** join/#asterisk gchowdhury (4531720a@gateway/web/freenode/ip.69.49.114.10) |
| 17:00.47 | *** join/#asterisk Micc (~Micc@c-98-232-41-66.hsd1.wa.comcast.net) |
| 17:02.06 | Micc | now that I'm able to see locks with core show locks, is there anything I should be looking for while asterisk is running normally? I see sometimes when running core show locks that it has one or two. I suppose this is normal operation and I just need to wait till something hangs to find the juicy stuff. |
| 17:02.22 | *** join/#asterisk ixyd_ (~denzs@ip-78-94-178-44.unitymediagroup.de) |
| 17:04.10 | ixyd_ | after updating my asterisk system from 1.4 to 1.8 the dialplan reload seems to take much longer than before, after looking in the full log it seems as a mysql lookup is done for every hint based on a sip peer, has this changed since 1.4 and if yes, what is the reason? :) |
| 17:04.22 | ixyd_ | 1.4 took about a few seconds and 1.8 takes about minutes... :( |
| 17:04.48 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
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| 17:34.22 | gchowdhury | is the syntax correct to add two string together and pass in to MD5 function Set(FILENAME=MD5(${EPOCH}${CHANNEL})) |
| 17:39.13 | Kobaz | ${MD5(... |
| 17:39.24 | Kobaz | any time you want a result from a function, ${FUNCTION(.. |
| 17:40.46 | leifmadsen | see the dialplan chapters in the asterisk book |
| 17:40.48 | leifmadsen | ~thebook |
| 17:40.48 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 17:44.37 | paulc | @leifmadsen: good webinar :) |
| 17:44.43 | leifmadsen | thanks! |
| 17:44.48 | leifmadsen | hope it was informative |
| 17:51.14 | gchowdhury | thanks |
| 17:51.59 | gchowdhury | does asterisk uses process or thread and is there a way to get the process or thread id |
| 17:52.11 | leifmadsen | asterisk is multi-threaded |
| 17:52.34 | gchowdhury | how do i get the thread id |
| 17:52.43 | leifmadsen | sounds like a linux question (I'm not sure) |
| 17:54.30 | cj | what tools can I use to verify that inbound PSTN calls are picked up correctly to the dialplan? It seems that the s extension is not getting triggered |
| 17:55.32 | leifmadsen | look at the CDR() ? |
| 17:55.59 | cj | sorry, I'm still kinda' new. Can you tell me where I can read about the CDR()? |
| 17:57.10 | leifmadsen | core show function CDR |
| 17:57.12 | leifmadsen | or |
| 17:57.14 | leifmadsen | ~thebook |
| 17:57.14 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 17:57.58 | cj | great, thanks. I bought a copy of the 3rd edition and will look through the pdf |
| 18:00.06 | cj | okay, I can tell that they are not being answered... |
| 18:00.10 | cj | but I want to determine why... |
| 18:00.15 | Kobaz | gchowdhury: you need to compile with DEBUG_THREADS, and use core show threads |
| 18:00.16 | *** join/#asterisk radic (~radic@dslb-178-002-236-148.pools.arcor-ip.net) |
| 18:00.25 | Kobaz | gchowdhury: but you won't get the LWP... but i have a patch that can do that |
| 18:00.29 | cj | how can I determine where calls from DAHDI channel 4 should go, and how do I update them? |
| 18:00.48 | cj | I've set the default context to 'default' and added this to the dialplan... |
| 18:01.17 | cj | exten => s,1,Answer() same => n,Playback(tt-weasels) same => n,Hangup() |
| 18:01.25 | *** join/#asterisk CodyG (~cody@216-102-15-215.scoe.org) |
| 18:01.28 | *** join/#asterisk fauxalliance (~fauxallia@142.163.150.199) |
| 18:02.24 | CodyG | Anyone successfully get gtalk to work with Asterisk? |
| 18:02.36 | cj | I've gotten my own xmpp server to work |
| 18:02.37 | *** join/#asterisk species86 (~Species@c053880-02-v1031-static.nvrninaa.cinergymetronet.net) |
| 18:02.42 | paulc | cj: do a "core set debug 5" and "core set verbose 5" and see if the console gives you any clues as to what's going on when an inbound call arrives |
| 18:02.50 | cj | great, thanks |
| 18:03.02 | species86 | im having issues getting asterisk to run a shell command |
| 18:03.08 | species86 | i want to dial an extension |
| 18:03.16 | species86 | and have that extension auto answer |
| 18:03.23 | species86 | then run a shell script that plays an audio file |
| 18:03.32 | species86 | i have the shell script working |
| 18:03.51 | species86 | but i can't get the system() command to work |
| 18:03.54 | CodyG | After installing I get: OpenSSL not installed. You need to install OpenSSL on this system, or disable the TLS option in your configuration file....I have openssl/gnutls installed, jabber.conf |
| 18:04.15 | species86 | exten => 555,3,2,System(/usr/local/firealarm.sh) |
| 18:04.20 | CodyG | oops jabber.conf I have usetls=yes, am I supposed to change this to something else |
| 18:04.24 | species86 | anybody have a suggestion? |
| 18:04.47 | species86 | the whole thing looks like this |
| 18:04.49 | species86 | exten => 555,1,Answer |
| 18:04.49 | species86 | exten => 555,2,Wait(2) |
| 18:04.49 | species86 | exten => 555,3,2,System(/usr/local/firealarm.sh) |
| 18:04.49 | species86 | exten => 555,4,Hangup |
| 18:05.31 | cj | does this make any sense to anyone? It seems that it's directing the call correctly, but the line is not picked up: http://paste2.org/p/1376340 |
| 18:07.11 | *** join/#asterisk AlxCds (d1d04401@gateway/web/freenode/ip.209.208.68.1) |
| 18:07.27 | species86 | no one has experience with system()? |
| 18:08.14 | *** join/#asterisk theron (~Theron@ip244.scolloc.lh.net) |
| 18:08.57 | c4rg | species86: exten => 555,3,System... |
| 18:09.02 | c4rg | why there's '2' ? |
| 18:09.10 | c4rg | also is this script executable? |
| 18:09.18 | species86 | yes it is executable |
| 18:09.42 | c4rg | why is there a '2' ? exten => 555,3,2,System(/usr/local/firealarm.sh) |
| 18:09.43 | species86 | i've even logged in as the asterisk user to make sure it can run it |
| 18:10.10 | species86 | good question |
| 18:10.13 | species86 | let me try again |
| 18:10.21 | c4rg | try to remove it and reload the dialplan |
| 18:10.42 | species86 | yea |
| 18:10.42 | species86 | nothing |
| 18:11.11 | species86 | it just connects |
| 18:11.14 | species86 | waits 2 seconds |
| 18:11.16 | species86 | and then it hangs up |
| 18:11.29 | species86 | but the script never runs |
| 18:14.06 | c4rg | what this script is supposed to do? |
| 18:14.43 | species86 | it just plays an ogg file |
| 18:14.47 | species86 | out the sound card |
| 18:14.54 | species86 | i have the sound card plugged into the pa system |
| 18:15.00 | species86 | basically it's to play a fire alarm tone |
| 18:15.06 | Kobaz | you should be using TrySystem |
| 18:15.14 | Kobaz | it gives you more data |
| 18:15.18 | c4rg | perhaps you could run something like system( |
| 18:15.29 | c4rg | system(/usr/local/firealarm.sh &> /tmp/log) |
| 18:15.38 | c4rg | to redirect the output to /tmp/log |
| 18:15.52 | c4rg | you could then see if it runs and possibly where this command fails |
| 18:16.10 | species86 | ok |
| 18:16.12 | species86 | let me try |
| 18:16.39 | species86 | interesting |
| 18:16.53 | species86 | it looks like it's a soundcard issue |
| 18:16.55 | species86 | like |
| 18:16.58 | species86 | a soundcard permission issue |
| 18:17.02 | c4rg | yeah |
| 18:17.09 | species86 | ALSA lib confmisc.c:768:(parse_card) cannot find card '0' |
| 18:17.10 | species86 | ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or direct$ |
| 18:17.10 | species86 | ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings |
| 18:17.10 | species86 | ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory |
| 18:17.10 | species86 | ALSA lib confmisc.c:1251:(snd_func_refer) error evaluating name |
| 18:17.12 | species86 | ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory |
| 18:17.15 | species86 | ALSA lib conf.c:3985:(snd_config_expand) Evaluate error: No such file or directory |
| 18:17.15 | c4rg | probably asterisk can't access the device |
| 18:17.17 | species86 | ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM default |
| 18:17.19 | species86 | sox: Failed writing default: cannot open audio device |
| 18:17.28 | species86 | what's weird is i can execute the script |
| 18:17.31 | species86 | if i login as user asterisk |
| 18:18.02 | c4rg | try this: sudo -u asterisk /usr/local/firealarm.sh |
| 18:18.58 | species86 | Sorry, user asterisk is not allowed to execute '/usr/local/firealarm.sh' as asterisk on trixbox1.localdomain. |
| 18:19.19 | c4rg | did you run this command as root user? |
| 18:19.45 | CodyG | argh...anyone know where to start on a fix for this issue? res_jabber.c:1610 aji_act_hook: OpenSSL not installed. You need to install OpenSSL on this system, or disable the TLS option in your configuration file |
| 18:19.59 | species86 | i logged in as asterisk |
| 18:20.00 | species86 | and ran it |
| 18:25.08 | c4rg | species86: no, try to run the command mentioned above as root |
| 18:25.35 | species86 | same thing |
| 18:25.41 | species86 | [trixbox1.localdomain ~]# ALSA lib confmisc.c:768:(parse_card) cannot find card '0' |
| 18:28.59 | c4rg | could you paste somewhere the output of ps axu | grep asterisk ; id asterisk ; ls -alh /dev/snd |
| 18:29.02 | c4rg | ? |
| 18:29.19 | species86 | i think i just fixed it |
| 18:29.21 | species86 | let me check |
| 18:29.30 | species86 | i had to add asterisk to a group |
| 18:29.34 | species86 | but now when i run that command |
| 18:29.35 | species86 | it works |
| 18:29.48 | *** join/#asterisk _jdccdevel (~chatzilla@taurus.whctcomm.net) |
| 18:29.59 | c4rg | :) |
| 18:30.03 | species86 | although |
| 18:30.05 | species86 | it's still not working |
| 18:30.07 | species86 | when i dial |
| 18:30.07 | species86 | lol |
| 18:30.25 | c4rg | paste the output |
| 18:30.32 | species86 | of that command? |
| 18:30.38 | c4rg | ps axu | grep asterisk ; id asterisk ; ls -alh /dev/snd |
| 18:31.03 | species86 | give me a sec |
| 18:31.08 | _jdccdevel | Has anyone had experience using the Asterisk::LCR perl module from CPAN with Asterisk 1.6 ? |
| 18:31.42 | species86 | http://www.heypasteit.com/clip/WOS |
| 18:32.18 | species86 | didn't get everything |
| 18:32.19 | species86 | http://www.heypasteit.com/clip/WOT |
| 18:32.21 | species86 | there we go |
| 18:35.08 | c4rg | oh |
| 18:35.23 | c4rg | restart asterisk |
| 18:37.55 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 18:38.29 | *** join/#asterisk BMJ (~bjohns@c-24-126-158-110.hsd1.ga.comcast.net) |
| 18:38.29 | *** mode/#asterisk [+o BMJ] by ChanServ |
| 18:39.41 | species86 | give me a second |
| 18:39.45 | species86 | im just going to restart the whole machine |
| 18:43.12 | species86 | ok |
| 18:43.15 | species86 | restarted and still nothing |
| 18:43.38 | species86 | same error in the log |
| 18:44.07 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 18:44.10 | atan | Anyone recall what the line for a cisco ip phone config file is to change the unlock password? |
| 18:47.00 | species86 | sweet |
| 18:47.02 | species86 | i fixed it |
| 18:47.10 | species86 | changed some of the permissions on my sound card |
| 18:47.15 | species86 | and that fixed it |
| 18:47.17 | species86 | just set them to 777 |
| 18:47.18 | species86 | lol |
| 18:47.23 | species86 | not that worried about it for this box |
| 18:47.28 | species86 | thanks for your help |
| 18:47.54 | *** join/#asterisk TimeRider (~steve@188.220.33.123) |
| 18:59.03 | *** join/#asterisk mclaro (~mclaro@201.250.84.162) |
| 19:03.09 | *** join/#asterisk fofware (~Fabian@host5.186-124-226.telecom.net.ar) |
| 19:10.41 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 19:11.06 | *** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:21f:5bff:fe37:c2c9) |
| 19:27.55 | iprouteth0 | would anyone be able to help with this http://pastebin.com/JtEGcpTL google voice outbound trouble |
| 19:29.00 | *** join/#asterisk bent_screwdriver (~bent_scre@74.255.249.66) |
| 19:30.07 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 19:35.45 | drfreeze | I installed g729 codec yesterday. We are using Zoiper Business (with g729) |
| 19:36.26 | drfreeze | I'm now getting pages of translation errors when trying to connect |
| 19:36.26 | drfreeze | http://pastie.textmate.org/private/sscshzmycuij0z3vimsg |
| 19:42.01 | pabelanger | drfreeze: it tells you your issue: [Apr 21 14:33:36] WARNING[31546]: codec_g729a.c:499 get_license: out of G.729 decoder licenses |
| 19:42.57 | drfreeze | pabelanger: not sure what that means. Can you help? |
| 19:44.09 | pabelanger | drfreeze: you need more g729 licenses |
| 19:45.35 | drfreeze | I only have one channel in use. Do I need 2 licenses? |
| 19:48.38 | pabelanger | drfreeze: Depends, why are you transcoding wav to g729? install g729 prompts |
| 19:49.30 | drfreeze | pabelanger: not sure why it is doing that. I need to look up g729 prompts. Don't know what that is. |
| 19:49.55 | drfreeze | oh, maybe it is recording the call |
| 19:51.29 | pabelanger | drfreeze: yes, you need to figure out why asterisk is transcoding first, then figure out how to resolve your issue |
| 19:53.04 | drfreeze | or, just buy new licenses |
| 19:56.29 | leifmadsen | recording the call will certainly use more licenses than 1 |
| 19:56.35 | leifmadsen | you'll probably need 3 per call |
| 19:57.09 | trintiron | leifmadsen are you from denmark ? |
| 19:58.17 | leifmadsen | https://issues.asterisk.org/view.php?id=13513 |
| 19:58.21 | leifmadsen | trintiron: no, Canada |
| 19:58.31 | leifmadsen | pabelanger: more info above for you |
| 19:59.07 | trintiron | okay it is just a typical danish name |
| 19:59.42 | leifmadsen | yes it is :) |
| 19:59.50 | leifmadsen | (my late grandfather was from Copenhagen) |
| 20:01.50 | trintiron | okay thats why |
| 20:02.18 | leifmadsen | :) |
| 20:02.37 | leifmadsen | I pronounce it "Life" |
| 20:03.13 | pabelanger | leifmadsen: indeed, yes. more licenses for recording. licenses for each stream |
| 20:03.30 | tzanger | everyone else pronounces it "dickhead" |
| 20:03.45 | leifmadsen | tzanger: or Richard for short |
| 20:03.49 | tzanger | lawl |
| 20:04.07 | tzanger | I have no idea how I"ve got such a finely-tuned sense of when to flip over to this channel to rip on my friend |
| 20:04.10 | tzanger | but it's uncanny |
| 20:04.19 | leifmadsen | ears be on fire |
| 20:04.31 | leifmadsen | and... you have friends? |
| 20:04.36 | tzanger | ooh |
| 20:04.37 | tzanger | burn |
| 20:06.25 | fofware | hello guys, I'm on asterisk 1.8 now and lua_pbx do not see context created in extensions.conf and can't see contex created in extensions.lua from context in extensions.conf, any one know how I can see context created in extensions.conf from pbx_lua and if I can use switch statement in lua_pbx, and how and where use it? |
| 20:06.54 | tzanger | leifmadsen: out of all your cookbook recipes which are your favourites? |
| 20:07.53 | Kobaz | chocolate chip cookies |
| 20:10.55 | leifmadsen | tzanger: I like the hot-desk recipe :) |
| 20:11.07 | leifmadsen | the authentication ones are also neato |
| 20:11.20 | leifmadsen | and of course Making Grandma Louder and Growing Your Company With PITCH_SHIFT() |
| 20:13.04 | tzanger | haha |
| 20:13.13 | tzanger | I have to start playing with SIP/SIMPLE |
| 20:14.05 | tzanger | anveo has *canadian* dids with 2-way SMS |
| 20:14.12 | tzanger | hell they have kitchener dids |
| 20:14.33 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 20:21.54 | leifmadsen | tzanger: interesting! |
| 20:22.15 | BMJ | Asterisk SCF Developer call at 5:00 Eastern. Info here: https://wiki.asterisk.org/wiki/display/TOP/Asterisk+SCF+Developer+Call+-+04212011+-+1700+EST |
| 20:28.31 | *** join/#asterisk path (~luis@201.238.249.133) |
| 20:29.14 | path | hey guys Im trying to get the callerid through ${CALLERID |
| 20:29.16 | path | } |
| 20:29.47 | path | but it just outputs nothing. The call is being done in a ZAP Channel |
| 20:37.25 | *** join/#asterisk bent_screwdriver (~bent_scre@74.255.249.66) |
| 20:38.30 | path | :( |
| 20:40.18 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 20:40.28 | *** join/#asterisk slugs_ (~chatzilla@unaffiliated/slugs) |
| 20:40.57 | slugs_ | Hey all |
| 20:42.25 | slugs_ | I have polycom IP 321's and im trying to delete a user out of the 'on screen' directory and can't seem to find where this entry is being stored |
| 20:42.30 | slugs_ | any help would be appreciated |
| 20:47.24 | Kobaz | on the tftp/ftp server in the overrides.xml files |
| 20:48.48 | slugs_ | Kobaz: I don't have that file in there |
| 20:49.01 | slugs_ | just all the mac-directory.xml's |
| 20:50.30 | *** join/#asterisk m4xx (~m4xx@c-76-19-95-158.hsd1.ct.comcast.net) |
| 20:51.58 | Kobaz | then it's not stored |
| 20:52.04 | Kobaz | you can reboot the phone and it will be gone |
| 20:52.25 | slugs_ | i tried... |
| 20:52.28 | Kobaz | er |
| 20:52.29 | Kobaz | i mean |
| 20:52.30 | slugs_ | it's still there |
| 20:52.40 | Kobaz | er... it's in directories, not overrides |
| 20:53.43 | slugs_ | mac-directory.xml? |
| 20:54.00 | Kobaz | yeah everything is by mac |
| 20:54.48 | slugs_ | yeah what i did was 'mv mac-* /tmp |
| 20:55.03 | slugs_ | rebooted the phone, and still see it's listing |
| 20:55.28 | Kobaz | grep your dir for the item in question |
| 20:55.33 | Kobaz | it's in there somewhere |
| 20:55.43 | Kobaz | you may have it hard coded in sip.cfg or something similar |
| 20:55.48 | *** join/#asterisk m4xx (~m4xx@c-76-19-95-158.hsd1.ct.comcast.net) |
| 20:56.06 | m4xx | anyone know how to do a factory reset on a cisco spa501g by any chance? =\ |
| 20:58.22 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 21:00.44 | slugs_ | damn can't seem to find this damn file |
| 21:01.14 | slugs_ | root@asterisk:/tftproot# grep -iR 0004f2a5c123 |
| 21:03.17 | *** join/#asterisk cerberus_za (~coert@196-210-190-65.dynamic.isadsl.co.za) |
| 21:05.06 | *** join/#asterisk luisfelice (~luisfelic@186.188.77.180) |
| 21:07.02 | luisfelice | Hi All, I am trying to configure the asterisk 1.8 as Google Talk, but the CLI returns: JABBER: socket read error. I installed the libiksemel-dev and libssl-dev. I am using debian 6 |
| 21:07.16 | *** join/#asterisk m4xx (~m4xx@c-76-19-95-158.hsd1.ct.comcast.net) |
| 21:07.19 | luisfelice | Any idea? |
| 21:07.27 | Micc | now that I have debug fd leaks on I can see that udptl sockets are getting created for every call, even ones without t38pt_udptl set, and even peers with t38pt_udptl=no. This seems like a waste of resources to me. |
| 21:10.27 | luisfelice | anyone? |
| 21:11.05 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 21:12.35 | nix8n82 | luisfelice, nat/firewall? |
| 21:12.56 | slugs_ | Kobaz: I see it in all the *directory.xml files! |
| 21:12.58 | *** join/#asterisk devil_evoxxx (~d3v1l@host228-97-dynamic.16-79-r.retail.telecomitalia.it) |
| 21:13.04 | luisfelice | nix8n82: no directly to internet, the server has a pluc IP |
| 21:13.06 | slugs_ | it's on each persons phone |
| 21:13.07 | luisfelice | public IP |
| 21:13.55 | devil_evoxxx | hi all, i'm having this warning on asterisk [Apr 21 23:10:04] WARNING[3180]: sig_pri.c:985 pri_find_dchan: Span 4: No D-channels available! Using Primary channel as D-channel anyway! |
| 21:13.55 | slugs_ | Kobaz: is there an easy way to modify 20+ phones with the directory listing? |
| 21:15.46 | Kobaz | slugs_: sed |
| 21:15.51 | Kobaz | slugs_: or perl |
| 21:16.05 | keith4 | slugs_: it's a pain to get stuff out of polycom listings, once it's in there. the phones keep saving their conf when you reboot them. |
| 21:16.15 | keith4 | taking away write access to tftp, temporarily, might help |
| 21:17.45 | slugs_ | yea |
| 21:21.01 | slugs_ | thank you so much |
| 21:21.08 | devil_evoxxx | there's someone that have a sampel configuration of chan_dadhi.conf |
| 21:21.09 | devil_evoxxx | ? |
| 21:21.16 | *** join/#asterisk luisfelice (~luisfelic@186.188.77.180) |
| 21:22.02 | *** join/#asterisk LedZeplin (jbearer@74.46.1343.static.theplanet.com) |
| 21:22.38 | keith4 | devil_evoxxx: isn't there a tool to generate that? |
| 21:22.51 | fofware | I can't see context created in extensions.conf in extensions.lua, any one can help me? |
| 21:23.05 | LedZeplin | probably a simple question but I can't find the answer, how is it determined whiat channels are in a outbound trunk? |
| 21:24.58 | LedZeplin | or I should say a trunk that is used by an outbound route |
| 21:25.52 | *** join/#asterisk luisfelice (~luisfelic@186.188.77.180) |
| 21:26.56 | luisfelice | sorry I was out of battery, anyone have an idea of my problem? |
| 21:28.11 | *** part/#asterisk luisfelice (~luisfelic@186.188.77.180) |
| 21:31.33 | Micc | msg luisfelice, I haven't tried jabber with 1.8, but it works great in 1.6, but I do see that error all the time, I just ignore it, it still works. |
| 21:33.13 | LedZeplin | ah, group=0, duh. |
| 21:33.17 | *** part/#asterisk LedZeplin (jbearer@74.46.1343.static.theplanet.com) |
| 21:36.38 | *** join/#asterisk cyford (~cyford@96-25-169-243.gar.clearwire-wmx.net) |
| 21:42.17 | devil_evoxxx | keith, what kind of tool ar you taling about? dadhi_config? |
| 21:49.54 | devil_evoxxx | p.s. i'm not using freepbx |
| 21:51.07 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3.3 (2011/04/21), 1.6.2.17.3 (2011/04/21), 1.4.40.1 (2011/04/21), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.1 (2011/04/01), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 21:51.32 | leifmadsen | Asterisk security releases for the 1.4, 1.6.1, 1.6.2, and 1.8 branches are now available. See the release announcement for more information: http://www.asterisk.org/node/51622 |
| 22:01.39 | *** part/#asterisk AlxCds (d1d04401@gateway/web/freenode/ip.209.208.68.1) |
| 22:15.00 | Freeaqingme | leifmadsen, tnx |
| 22:15.17 | Freeaqingme | leifmadsen, is there a security mailinglist or alike to stay up2date? |
| 22:15.41 | Qwell | Freeaqingme: http://lists.digium.com/mailman/listinfo/asterisk-announce |
| 22:15.48 | Freeaqingme | kk, tnx |
| 22:15.58 | Qwell | http://lists.digium.com/mailman/listinfo/asterisk-security too |
| 22:20.13 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 22:35.04 | *** join/#asterisk hacktu (~Aaz@pantheon.rus-com.net) |
| 22:36.34 | *** join/#asterisk mclaro (~mclaro@201.250.84.162) |
| 22:40.37 | *** join/#asterisk fhmiv (~fhmiv@65.5.242.247) |
| 22:43.12 | Freeaqingme | wtf, I'm watching the astricon '10 keynote, and the ceo of nswitch says they're using agi for billing. Isn't that a weird choice? |
| 22:57.47 | citywok | i've got an 800# that when i dial sends media right after the 183, and * doesn't figure that out. how do i get * to pick that up? |
| 23:06.55 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 23:12.17 | citywok | Alright, here's the graph from wireshark of a call that rings in to oblivion, even though they start sending me media (They being AT&T), how do i make asterisk listen to this? http://citywok.me/files/files/1/asteriskerror.png |
| 23:21.04 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 23:22.56 | Freeaqingme | How likely is it for Asterisk to be merged with Asterisk SCF with asterisk 2? |
| 23:33.08 | *** join/#asterisk qubix (~q@unaffiliated/qubix) |
| 23:37.49 | pabelanger | Freeaqingme: Asterisk and Asterisk SCF are 2 different projects. 2 different development teams working in parallel. |
| 23:37.54 | pabelanger | ~collectdebug |
| 23:37.54 | infobot | collectdebug is, like, a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 23:38.07 | *** join/#asterisk jstapleton (~jstapleto@c-24-125-171-223.hsd1.va.comcast.net) |
| 23:38.08 | pabelanger | citywok: ^ pb the debug log. Lets see what is going on |
| 23:38.30 | Freeaqingme | pabelanger, yes, but because the ultimate goal (providing telephony) is the same, I could imagine the two are merged at some time |
| 23:39.10 | pabelanger | Freeaqingme: Merge no. Ability to connect both together, yes |
| 23:39.57 | pabelanger | Freeaqingme: EG: Asterisk SCF will not provide conferencing, asterisk does. |
| 23:40.03 | Freeaqingme | kk |
| 23:40.25 | pabelanger | But Asterisk SCF could be setup to utilize Asterisk conference abilityes |
| 23:41.12 | Freeaqingme | yeah, I just watched the astricon keynote where that was demonstrated |
| 23:49.37 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 23:54.58 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 23:57.27 | *** join/#asterisk wonderworld (~ww@port-92-201-93-172.dynamic.qsc.de) |
