| 00:00.55 | deweydb | nice that was it |
| 00:00.57 | deweydb | thanks |
| 00:01.38 | leifmadsen | np, glad you figured it out so quickly :) |
| 00:02.08 | *** join/#asterisk rx4change (~joshelson@216.241.45.82) |
| 00:02.38 | rx4change | Has anyone ever seen this error "retrieve_file: SQL Get Data error! coltitle=category" with Asterisk 10 realtime voicemail? |
| 00:02.53 | rx4change | It's happening on the query "SELECT * FROM voicemail_messages WHERE dir=? AND msgnum=?" |
| 00:03.16 | rx4change | I believe it's the culprit on a frequent deadlock condition I'm seeing on Asterisk too |
| 00:03.21 | rx4change | seems to be a bit leaky |
| 00:03.34 | leifmadsen | odbc? |
| 00:03.38 | rx4change | correct |
| 00:03.51 | leifmadsen | what version? |
| 00:04.04 | rx4change | 10.9.0 vanilla |
| 00:04.12 | leifmadsen | of unixODBC I mean |
| 00:04.39 | rx4change | oh... installed from RPM on fedora 2.2.14-13 |
| 00:07.08 | rx4change | possibly a version issue there? |
| 00:08.59 | dencon | Have you had a look at this: http://forums.asterisk.org/viewtopic.php?f=1&t=81495 |
| 00:09.17 | rx4change | Yeah... didn't quite match up to what I was seeing |
| 00:09.34 | rx4change | my backend is MySQL and the "category" field does exist in the DB |
| 00:09.34 | deweydb | anyone know why i keep getting this error message over and over again in my CLI |
| 00:09.35 | deweydb | http://pastebin.com/Fp1rK8UC |
| 00:10.08 | rx4change | @deweydb - something is trying to authenticate to your manager |
| 00:10.15 | rx4change | you running a HUD server or something like that? |
| 00:10.38 | leifmadsen | rx4change: CentOS? |
| 00:10.45 | leifmadsen | because what ships with CentOS is from like... 2005 :) |
| 00:10.53 | leifmadsen | it's suggested you install unixODBC from source |
| 00:10.55 | leifmadsen | unixodbc.org |
| 00:11.02 | rx4change | Haha... I'm running Fedora 16-ish |
| 00:11.05 | rx4change | it's still not new |
| 00:11.08 | rx4change | looks like 2.3.1 is current |
| 00:11.13 | rx4change | I'll build from source and see if if I goes away |
| 00:11.18 | rx4change | that's a good call Leif |
| 00:13.01 | dencon | I'm trying to debug a segfault but I don't have an un-optimized build available unfortunately, so I have some "value optimized out" in gdb. I know where it crashes, and looking at the code... |
| 00:13.03 | dencon | Line 459 - 463: http://doxygen.asterisk.org/asterisk1.8/chan__agent_8c-source.html |
| 00:13.16 | dencon | Is that actually doing anything? It looks to me it's just going through a list and if !strcmp(p->agent, agt) it just breaks. Am I missing anything? |
| 00:14.24 | deweydb | HUD server? |
| 00:14.47 | deweydb | i am running freePBX and asterisk. |
| 00:15.02 | deweydb | there are only two extensions and it all goes to me, nobody else is using the lines |
| 00:15.23 | deweydb | seems to be trying to authenticate every 10 seconds. |
| 00:15.32 | rx4change | yeah.. you have some process that's trying to do it |
| 00:15.55 | deweydb | strange. ok. well im not every sure where to start looking for that. |
| 00:16.12 | deweydb | is this default behaviour? because i haven't installed that much other than just asterisk and freepbx |
| 00:16.30 | rx4change | that isn't Asterisk itself |
| 00:16.32 | navaismo | deweydb, freepbx use the manager too check your amportal.conf with your manager.conf |
| 00:16.37 | rx4change | something local to the box is trying to login |
| 00:17.15 | deweydb | so i ahve the wrong username in my freepbx settings somewher |
| 00:17.19 | deweydb | ok will look for that |
| 00:17.26 | deweydb | any idea what file that would be in? |
| 00:17.36 | rx4change | depends on version |
| 00:17.57 | rx4change | new stuff is in DB, I believe |
| 00:17.58 | deweydb | 2.10.0.0 (freepbx)( |
| 00:18.00 | rx4change | old stuff is in amportal.conf |
| 00:18.03 | rx4change | iirc |
| 00:18.31 | navaismo | deweydb, manager.conf and check the user and pass in your GUI if doesnt match change it |
| 00:20.10 | Orii | asterisk.org is down? |
| 00:20.27 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 00:21.16 | navaismo | Orii, nope i can open www.asterisk.org |
| 00:21.29 | file | Orii, works here |
| 00:21.44 | Orii | huh.... |
| 00:21.54 | file | what happens? |
| 00:22.01 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 00:22.04 | Orii | unable to connect screen in firefox |
| 00:22.12 | file | weird |
| 00:22.13 | jpsharp | https://asterisk.org doesn't work. |
| 00:22.44 | jpsharp | but regular not https does. Somethings redirect to the SSL side. |
| 00:22.46 | Orii | i used google to open the iste |
| 00:22.53 | Orii | site* |
| 00:23.01 | Orii | and i just tried again with the link navaismo made |
| 00:23.04 | Orii | still nothing |
| 00:23.51 | Orii | does the site have any problems with privoxy? |
| 00:24.29 | deweydb | navaismo: http://pastebin.com/6fxcVnzE (i commented this block out before because i thought it looked dangerous to have a default user and default password in manager.conf |
| 00:24.36 | deweydb | and that was the problem. |
| 00:24.44 | deweydb | i put it back. but now i am still a little worried about that. |
| 00:26.27 | navaismo | deweydb, then change it using the gui |
| 00:27.20 | deweydb | navaismo: yeah, i figured, but i can't find that setting anywhere. |
| 00:27.54 | navaismo | deweydb, in advanced settings |
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| 00:29.02 | Orii | hmm digium works at least |
| 00:29.05 | navaismo | set to true the advanced settings details apply an you will see the manager's user & pwd |
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| 00:31.26 | deweydb | navismo: the username 'user' is nowhere in that settings page. |
| 00:31.37 | deweydb | sorry: navaismo* |
| 00:32.02 | navaismo | did you have set to TRUE the advanced settings details? |
| 00:32.28 | deweydb | yes, and hidden, and i even enabled readonly because i thought it might behidden |
| 00:32.44 | deweydb | i see this field: User Portal Admin Username |
| 00:32.50 | deweydb | but that is for loggin in to freepbx gui |
| 00:34.57 | navaismo | deweydb, I got to go but apply config and then you will see the seetings this is how i see it http://i.imgur.com/itNGS.png |
| 00:41.58 | deweydb | ah thanks |
| 00:43.55 | deweydb | don't know how i missed that. |
| 00:43.57 | deweydb | thanks |
| 00:51.23 | volga629 | IAX trunk issue can't resolve. on remote sip extension rtp is not working http://fpaste.org/cWYU/ |
| 00:51.33 | volga629 | any help thank you |
| 00:51.53 | *** join/#asterisk Nemus (~Nemus@c-67-161-251-38.hsd1.ut.comcast.net) |
| 00:52.12 | volga629 | IAX trunk between 2 asterisk boxes on another end sip extensions |
| 00:53.22 | *** join/#asterisk Sprocks (~Sprocks@bas1-brampton37-3096592814.dsl.bell.ca) |
| 00:53.45 | volga629 | also here some debug output of SDP http://fpaste.org/XLQ5/ |
| 00:55.56 | *** join/#asterisk ruben231 (~OpenDial@112.198.77.162) |
| 01:04.37 | volga629 | chan_sip.c: Allocating new SIP dialog for 15c0294f7df8b3f711538f660f4982c2@ip:5060 - INVITE (No RTP) is this normal ? |
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| 01:38.25 | rx4change | leifmadsen: compiling from source on unixodbc 2.3.1 didn't change the problem |
| 01:38.38 | rx4change | still throwing this: [2012-10-16 19:37:17] WARNING[20357]: app_voicemail.c:3457 retrieve_file: SQL Get Data error! coltitle=category |
| 01:38.38 | rx4change | [SELECT * FROM voicemail_messages WHERE dir=? AND msgnum=?] |
| 01:56.14 | leifmadsen | rx4change: next thing to do then is to enable sql logging on the server (MySQL etc) and look at what Asterisk is requesting and finding out why it doesn't get what it expects |
| 01:56.16 | leifmadsen | goes to bed |
| 01:58.39 | rx4change | so I was able to get the error to go away by changing the default value of category from NULL to "" in MySQL |
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| 03:01.37 | john_roland_1359 | hello. my asterisk server keeps giving me this error message ---> http://pastebin.com/5zeYvbTs |
| 03:02.48 | john_roland_1359 | Person A can answer and receive a phone call from person B. But person B cannot receive a phone call from Person A |
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| 03:29.14 | ectospasm | john_roland_1359: is one of those peers missing allow=g729? |
| 03:29.37 | ectospasm | or does one of them have "disallow=all" and ONLY "allow=g729"? |
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| 03:39.19 | john_roland_1359 | ectospam: |
| 03:39.31 | john_roland_1359 | ectospam: 1 min. let me check |
| 03:41.24 | ectospasm | john_roland_1359: if you'd spell my nick right I'd be properly highlighted... |
| 03:41.44 | john_roland_1359 | ectospasm: sorry |
| 03:41.50 | ectospasm | tab completion ftw! |
| 03:42.06 | john_roland_1359 | ectospasm: ah yes. i can u se tab to complete |
| 03:42.09 | john_roland_1359 | use* |
| 03:42.28 | john_roland_1359 | ectospasm, :thanks. i am a little rusty on the protocols |
| 03:43.07 | ectospasm | for SIP codecs, I recommend disallow=all, and allow= only those codecs you want |
| 03:43.14 | john_roland_1359 | ectospasm, : only the general and my voip provider have allow |
| 03:44.03 | ectospasm | john_roland_1359: you can have a separate allow= for each codec, and unless you know what you're doing you can probably set everything in general and be done with it |
| 03:44.29 | ectospasm | john_roland_1359: you'd also need to set both phones A and B to use the same codec (at the phone side) |
| 03:44.38 | john_roland_1359 | ectospasm: [general] and [voip provider] have allow=ulaw and allow=g729 |
| 03:44.57 | ectospasm | john_roland_1359: have you enabled those codecs on the phones, too? |
| 03:45.13 | ectospasm | john_roland_1359: and G.729 is loaded, yes? |
| 03:45.24 | ectospasm | "g729 show licenses" |
| 03:45.25 | john_roland_1359 | ectospasm: on the phones. no. |
| 03:46.25 | ectospasm | ...basically it looks like something is trying to negotiate G.729 when there is no translation path |
| 03:46.26 | john_roland_1359 | ectospasm: so i think i should remove the codecs from the sip.conf |
| 03:46.39 | ectospasm | well, you'll need at least one codec |
| 03:47.07 | john_roland_1359 | ectospasm, :what would be the most basic codec that should work easily. |
| 03:47.40 | ectospasm | john_roland_1359: G.729 is patent-encumbered, you'll need a special codec module in order to use it. Digium sells one for 10USD per channel |
| 03:47.50 | ectospasm | I'd recommend ulaw |
| 03:48.01 | ectospasm | ...unless you need to save bandwidth |
| 03:48.06 | ectospasm | then go with gsm |
| 03:48.26 | ectospasm | ...there may be other free ones |
| 03:48.46 | john_roland_1359 | ectospasm: so i'll just have work with ulaw |
| 03:49.14 | ectospasm | don't forget the disallow=all line *above* allow=ulaw |
| 03:50.29 | john_roland_1359 | ectospasm, : do i set this for [general] , [voip-provider] and [peers] right? |
| 03:51.22 | *** join/#asterisk Naikrovek (~wwhite@unaffiliated/naikrovek) |
| 03:53.20 | ectospasm | john_roland_1359: if they're all the same, you don't need it in [provider] or [peers] |
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| 03:53.46 | john_roland_1359 | ok yes |
| 03:53.56 | john_roland_1359 | ectospasm, : you are right. it was the g729 |
| 03:55.54 | ectospasm | john_roland_1359: you're welcome! |
| 03:56.28 | john_roland_1359 | ectospasm: it works. i went to the [general] and [voip provider] and edited it as disallow=all and allow=ulaw |
| 03:56.41 | john_roland_1359 | ectospasm: thank you. i appreciate it. |
| 03:56.51 | ectospasm | no problem! |
| 03:57.05 | john_roland_1359 | :) |
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| 05:06.21 | jmls | what would cause a 7940 to stick at this : |
| 05:06.35 | jmls | 1 .1x REGISTERING 3600 125 [pbx-ip]:5060 |
| 05:07.30 | jmls | phone was in use, system crashed (hard reset on server), when server came back up this phone was no longer working. I've telnetted to it and verified all the information, all is right |
| 05:07.47 | jmls | phone can bin server, I can telnet to phone, so network is right |
| 05:07.59 | jmls | username and password match |
| 05:08.31 | ectospasm | rebooting the phone doesn't work? |
| 05:08.47 | ectospasm | sounds like the phone got its registration stuck |
| 05:08.50 | ectospasm | jmls: ^ |
| 05:09.11 | jmls | ectospasm: I've returned it to factory settings ;) |
| 05:09.34 | jmls | still no joy |
| 05:09.53 | ectospasm | are the phone's REGISTER packets getting to Asterisk? |
| 05:10.04 | ectospasm | ...check with tcpdump/wireshark |
| 05:10.11 | jmls | about to to that |
| 05:10.14 | jmls | ta |
| 05:11.28 | jmls | I'm getting register stuff hitting asterisk |
| 05:12.12 | ectospasm | is Asterisk responding appropriately, e.g. 407 Proxy Authentication Required, etc.? |
| 05:12.32 | jmls | no |
| 05:13.06 | jmls | every 2 seconds I get <--- SIP read from UDP:[phoneip]:[port] ---> |
| 05:13.20 | jmls | 10 times |
| 05:13.28 | ectospasm | pastebin your SIP debug |
| 05:14.44 | jmls | http://pastebin.com/uW6Kk67s |
| 05:18.41 | ectospasm | those all look like retransmissions... I'd have to see what the pcap said... if you turn up core debug and verbose, do you get any complaints from Asterisk? |
| 05:19.48 | jmls | core set debug 9 |
| 05:19.55 | jmls | core set verbose 10 |
| 05:20.07 | ectospasm | and the obligatory console => notice,warning,error,verbose,debug in logger.conf |
| 05:20.08 | jmls | gives nothing on the console except the sip debug |
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| 05:20.23 | jmls | oops |
| 05:20.26 | ectospasm | (be sure to run logger reload if you change logger.conf) |
| 05:20.29 | jmls | checks |
| 05:21.10 | jmls | yikes! |
| 05:21.16 | jmls | loads of debug messages |
| 05:21.27 | ectospasm | yep |
| 05:21.52 | jmls | you want a pastebin of this ;) |
| 05:22.04 | ectospasm | those tell us what Asterisk is really doing in the background... |
| 05:22.07 | ectospasm | ...so yes |
| 05:23.59 | jmls | before I do that: this looks suspicious : |
| 05:24.45 | jmls | [Oct 17 06:21:32] DEBUG[3879]: chan_sip.c:9071 find_call: REGISTER request has no from tag, dropping callid: 000cce74-038e0002-23b20774-6bd88cd4@phoneip from: sip:*0040*@serverIP |
| 05:24.45 | jmls | [Oct 17 06:21:32] DEBUG[3879]: chan_sip.c:28062 handle_request_do: Invalid SIP message - rejected , no callid, len 309 |
| 05:27.10 | ectospasm | those are just DEBUG messages, but it looks like Asterisk doesn't like the registration from the phone |
| 05:28.10 | jmls | if the registration message is rejected |
| 05:28.21 | jmls | no registration, right ;) |
| 05:29.27 | jmls | no warnings in the capture |
| 05:29.33 | jmls | just loads of debug |
| 05:29.46 | ectospasm | what version of Asterisk is this? |
| 05:30.08 | jmls | Asterisk SVN-branch-11-r374932 |
| 05:30.13 | jmls | almost latest |
| 05:30.32 | ectospasm | in production? |
| 05:30.41 | jmls | ... yes ... |
| 05:30.45 | ectospasm | tsk, tsk |
| 05:30.49 | jmls | heh |
| 05:30.53 | jmls | worked fine in testing |
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| 05:31.04 | ectospasm | production ain't testing, is it? (-; |
| 05:31.20 | jmls | we did nearly 100,000 calls, and were desperate to move from old |
| 05:31.41 | jmls | sometimes there's only one way to test and that's to go live |
| 05:31.49 | ectospasm | you may have uncovered a bug, I suggest you report it: https://issues.asterisk.org/jira |
| 05:32.07 | jmls | urgh |
| 05:32.15 | jmls | thanks for all the help |
| 05:32.46 | ectospasm | *especially* because this is a pre-11.0 release you're using |
| 05:34.43 | kaldemar | jmls: what was your previous version? |
| 05:34.49 | ectospasm | jmls: are there features of 11 that are must-have for your organization? |
| 05:35.16 | ectospasm | if you can't cite a single one, I'd recommend 1.8, since it's more mature |
| 05:41.22 | jmls | kaldemar: customised version of pre 1.4 :) |
| 05:42.07 | jmls | ectospasm: we're using the new confbridge a lot, and want to use the xmpp driver in 11 |
| 05:42.09 | kaldemar | jmls: do you have pedantic=no in sip.conf? the default for that was changed in 1.8.0 and it caused likewise behavior with some cisco phones. |
| 05:42.59 | jmls | no, default to pedantic=yes |
| 05:43.00 | kaldemar | in case you did not test that version earlier with the same phones. |
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| 05:43.29 | jmls | we've been using theses phones for 6 years |
| 05:43.38 | jmls | the _same_ phones |
| 05:43.45 | kaldemar | but not with 1.8.0 or newer? |
| 05:43.48 | jmls | no |
| 05:43.54 | jmls | tested with -11 |
| 05:44.02 | jmls | not _all_ the phones have this problem |
| 05:44.26 | jmls | I'll try pedantic = no |
| 05:44.55 | jmls | wow. works ! |
| 05:45.10 | jmls | kaldemar: thanks, man |
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| 06:47.56 | bulkorok | hi |
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| 07:10.51 | wdoekes | jmls: pedantic=no has more side-effects.. you may want to edit the source and allow only the reg without from-tag (or.. better yet, upgrade the firmware on those phones |
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| 08:04.01 | oquidave | hello, anyone used chan_dongle to create a gsm channel? |
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| 08:15.38 | oquidave | i can't listen to voice on the gsm side http://pastebin.com/03gVeYRg |
| 08:15.44 | oquidave | but the call goes through |
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| 08:58.27 | oquidave | <PROTECTED> |
| 08:59.05 | ectospasm | not I |
| 09:01.56 | oquidave | am failling to understand why i can't hear the voice on a gsm phone...the call goes through and runs through the dialplan smoothly, http://pastebin.com/03gVeYRg ...but for the parts of the dialplan where asterisk has to playback something, i can't hear a thing!!! |
| 09:01.57 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 09:02.20 | oquidave | could be because of some incompatible codecs maybe?? |
| 09:02.27 | ChannelZ | yes |
| 09:02.46 | ChannelZ | but I don't know anything about the Dongle channel driver either. What is on the other end of it? What kind of media does it expect? |
| 09:04.13 | oquidave | ChannelZ: it's a normal gsm phone. So i i call the usb 3G modem number and it drops to a certain context in the dialplan. all that executes fine except for the voice part |
| 09:05.31 | oquidave | ive tried with both chan_dongle drivers(http://code.google.com/p/asterisk-chan-dongle/) and chan_datacard(http://code.google.com/p/datacard/w/list) but similar problem |
| 09:06.04 | oquidave | according to the list, my modem version is compatible too ...it's a huawei e220 |
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| 09:41.30 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 09:41.35 | bombev | Hi all :) |
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| 09:42.40 | bombev | is there any templete how to create custom context because now My Incoming number goes to wrong IVR, my trunk using context=from-trunk right now |
| 09:44.02 | kaldemar | if you use plain asterisk, everything is pretty much custom. if you're using freepbx, please continue in #freepbx. |
| 09:44.38 | mirela666 | bombev: There is no template, maybe from some third-parties softvares |
| 09:44.39 | mirela666 | you can read and fid some examples in extensions.conf file |
| 09:48.07 | bombev | mirela666 now I am reading extensions.conf file also extensions additional file |
| 09:48.19 | bombev | ~pb |
| 09:48.19 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
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| 10:48.09 | mathi | hi |
| 10:49.17 | mathi | to enter the dialplan with Zoiper, I have to Dial "100": exten => 100,1,Goto(main-menu,s,1) |
| 10:49.33 | mathi | but how to receive calls from the PSTN ? what would be the extension name ? |
| 10:53.09 | kaldemar | depends on how you connect to PSTN. |
| 10:53.52 | WIMPy | Either the dialed number or s. |
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| 11:30.56 | DrDamnit | dahdi_hardware sees my TDM800P, but dahdi_genconf completely ignores it. What am I missing? |
| 11:31.51 | bombev | can someone help me: http://pastebin.ca/2246699 |
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| 11:34.58 | kaldemar | bombev: $[ "${CALLERID(name)}" = "" ] ? -> $["${CALLERID(name)}" = ""]? |
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| 11:47.13 | mirela666 | Can MixMonitor save recording in two file formats at once |
| 11:48.05 | DrDamnit | are there any issues with DAHDI and the 3.2.0-29-generic kernel? |
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| 11:53.01 | leifmadsen | mirela666: no, but you can copy to a new format after the fact with something like sox |
| 11:54.48 | *** part/#asterisk nappy (~dnepangue@58.71.19.178) |
| 11:56.12 | DrDamnit | Changed from dahdi 2.6.1 to the current 2.6 svn branch. No change. It sees the T1 card fine, but not the TDM800. The card is known good. We just upgraded this system from an older OS (Ubuntu 10 LTS) to this one (Ubuntu 12.04 LTS). Are there any known issues with the 3.2 kernel and the TDM800? |
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| 12:06.35 | mirela666 | leifmadsen: ok thx, that was plan B :) |
| 12:08.13 | mirela666 | leifmadsen: thought it could like VoiceMail to save wav,gsm... |
| 12:08.21 | leifmadsen | nope |
| 12:08.32 | leifmadsen | voicemail is he exception, not the rule |
| 12:09.31 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:11.21 | DrDamnit | modprobe wctdm24xxp triggers segmentation fault. http://pastebin.com/3tGSRzHr does the TDM800P just now work on kernel 3.2? |
| 12:12.30 | DrDamnit | just *not work on kernel 3.2? |
| 12:12.36 | leifmadsen | DrDamnit: it doesn't appear any one else is having issues, or knows for sure. I'd suggest you open an issue on the bug tracker and revert back to your working system. |
| 12:12.54 | DrDamnit | will do. |
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| 12:41.34 | volga629 | Hello Everyone, having trouble to call between 2 asterisk server which connected with IAX trunk and on each box extensions is SIP when calling from A to B they here me, but I am not http://fpaste.org/kBdQ/ |
| 12:42.53 | [TK]D-Fender | volga629, [2012-10-17 14:33:29] VERBOSE[23822] chan_sip.c: Reliably Transmitting (NAT) to 10.23.130.100:1025: |
| 12:43.05 | [TK]D-Fender | volga629, Contact: <sip:101@94.100.30.210:5060> |
| 12:43.25 | [TK]D-Fender | volga629, You care calling that SIP phone on a PRIVATE IP.... and telling them you are at a PUBLIC IP |
| 12:43.37 | [TK]D-Fender | viogYou have screwed your NAT settings up. Go fix them. |
| 12:44.02 | [TK]D-Fender | volga629, You have screwed your NAT settings up. Go fix them. |
| 12:46.02 | volga629 | this box has only public no private and NAT is default let me double check all settings |
| 12:48.00 | volga629 | nat=yes that all what have in sip.conf |
| 12:50.05 | *** join/#asterisk raub (~raub@ip70-171-42-89.ga.at.cox.net) |
| 12:52.21 | cusco | volga629: [2012-10-17 14:33:29] VERBOSE[23822] chan_sip.c: Reliably Transmitting (NAT) to 10.23.130.100:1025: -- is that a public IP? |
| 12:52.43 | DrDamnit | leifmadsen: Here was the fix (for future reference) we changed the order of the PCI cards in the slots. Not sure why that fixed it, but who cares... it did. :-) |
| 12:53.05 | leifmadsen | odd... :) |
| 12:53.15 | cusco | weird |
| 12:53.16 | leifmadsen | thanks for following up |
| 12:53.40 | volga629 | <PROTECTED> |
| 12:53.50 | leifmadsen | maybe they just had to be reseated? or maybe the searching of the PCI bus is different in the newer kernels... ? |
| 12:53.51 | leifmadsen | who knows :) |
| 12:54.21 | cusco | volga629: then it cannot comunicate via tunnel |
| 12:54.22 | volga629 | Box A is head office, SIP remote is second branch |
| 12:55.17 | DrDamnit | leifmadsen: here was our process: 1. re-seat. (no joy). 2. remove te122. (no joy). 3. Put TDM card in TE122's slot, then put the TE122 in the slot the tdm card was just in. and it worked. |
| 12:55.42 | volga629 | I know this not firewall, because tried shutdown and result the same |
| 12:56.17 | cusco | so you mean: |BOX1|-----(VPN)---->|BOX2|-----(Public)---->|SIP-PEER| ???? |
| 12:56.20 | [TK]D-Fender | Thinking that just shutting it down will "fix" this is insane |
| 12:56.55 | volga629 | no just for test to eliminate which part is not right |
| 12:57.40 | volga629 | yes almost because on public ip box terminates 2 tunnel one from branch and from head office |
| 13:00.12 | [TK]D-Fender | <volga629> SIP---->BOX AIAX2 (VPN TUNNEL) <----------> (VPN TUNNEL)IAX2 BOX B(VPN TUNNEL)<----->(VPN TUNNEL) SIP extension. BOX B have one interface which public only no private <- I also don't trust the FIRST leg of this call |
| 13:00.16 | *** join/#asterisk serafie (~erin@nat/digium/x-ablwxinuotoenvyk) |
| 13:02.18 | *** join/#asterisk keycruncher (~Adium@c-174-59-233-118.hsd1.pa.comcast.net) |
| 13:02.22 | volga629 | I asked add on box one interface and try set it to private and change behavior of ipsec tunnel, but currently have to find where the trap in current configuration |
| 13:02.58 | volga629 | one more |
| 13:03.55 | volga629 | From Box 2 trough vpn tunnel I can ping one of branch sip phones |
| 13:04.02 | volga629 | through |
| 13:05.07 | *** join/#asterisk dolcea-htpc (~dolcea@186.19.132.2) |
| 13:06.36 | volga629 | I noticed that on branch firewall ALG for SIP and RTP is enabled, going shutdown this |
| 13:09.16 | [TK]D-Fender | .... |
| 13:10.38 | volga629 | Contact: <sip:101@94.100:5060> contact line should sip peer ? |
| 13:13.48 | volga629 | is that should be like this 101 ? because this is extension form head office |
| 13:13.52 | volga629 | from |
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| 13:18.55 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:20.18 | volga629 | I just wonder which options are available for this case ? |
| 13:25.43 | *** join/#asterisk sbszulu (~dundubala@105.247.152.163) |
| 13:34.08 | volga629 | http://fpaste.org/Vtjm/ that call form head office side |
| 13:34.16 | volga629 | from |
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| 13:44.24 | volga629 | http://fpaste.org/jjFQ/ iS this look better ? |
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| 13:52.27 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 13:59.35 | volga629 | http://fpaste.org/Wtk2/ here another call |
| 14:02.22 | volga629 | have leave right now I will continue look on this issue |
| 14:02.41 | volga629 | Thank you Everyone for opinion and help |
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| 14:12.18 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:19.59 | hurdman | hi, i have had : "<ZOMBIE>, while fd is blocked by thread" during a just iax and call files test. asterisk 1.6.2.24, any idea of where i should take a look |
| 14:20.03 | hurdman | ? |
| 14:20.20 | Qwell | ~upgrade asterisk |
| 14:20.20 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 14:20.24 | hurdman | ( i have only seen very old bug into the google cash about that ) |
| 14:20.36 | Qwell | There's a reason you've only found very old bugs. |
| 14:20.55 | hurdman | Qwell: when i say "old", it's about 1.4** |
| 14:21.01 | Qwell | So, ancient. |
| 14:21.07 | hurdman | ( and 1.2 ) |
| 14:21.10 | Qwell | As ancient as your version of Asterisk. |
| 14:21.13 | coppice | the young bugs all moved away to a better life in the city |
| 14:21.32 | Qwell | coppice: Not Florida? |
| 14:22.46 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:22.56 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 14:23.01 | hurdman | Qwell: it's the version i have, and there's some modules i have developped for the 1.6 , and i have no time to redev for the new version now |
| 14:23.31 | Qwell | hurdman: Well, you're using something unsupported. When it breaks, you get to keep both pieces. |
| 14:23.42 | hurdman | snif |
| 14:24.17 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 14:24.39 | hurdman | so say "upgrade" it's to say "i can't help you because i don't now anything about that bug " ? |
| 14:25.19 | hurdman | ( the same bug seem's to be into the 1.8 ) |
| 14:32.48 | [TK]D-Fender | hurdman, the problem will not gt fixed in 1.6.X. Those branches are EOL |
| 14:32.49 | Qwell | No, it means that nobody cares enough to look into it. |
| 14:33.05 | Qwell | If you can reproduce it on 1.8, great. Report a bug. |
| 14:34.20 | hurdman | https://reviewboard.asterisk.org/r/1400/diff/1/ <= it looks like that, i'll backport the patch |
| 14:34.56 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 14:40.35 | *** join/#asterisk Jouva (~jouva@static-98-111-131-98.phlapa.fios.verizon.net) |
| 14:45.38 | Jouva | So I'm trying to understand a little more about T.38 since we're moving from PSTN to a VoIP provider and we have a fax line. What do I need to make sure of other than that our own in-house Asterisk server supports T.38? Is T.38 in itself the end-all solution for receiving faxes from everybody? |
| 14:46.11 | Jouva | And a lot of examples I see seem to have Asterisk handle the TIFF files. I assume this is so it can act as a virtual fax machine. What about our current setup with an actual fax machine? |
| 14:47.16 | carrar | Jouva, you'll probably need to force your faxing speed on your fax machines to 9600 |
| 14:47.30 | Jouva | Ok |
| 14:47.33 | carrar | You will also need a ATA that supports T.38 |
| 14:47.39 | Jouva | Right |
| 14:47.50 | carrar | But it should work fine |
| 14:47.52 | Jouva | Does my VoIP provider need to support it? I think that was my biggest concern |
| 14:47.57 | carrar | as long as you don't have NAT issues blocking you |
| 14:48.07 | carrar | as there are T.38 re-invites |
| 14:48.15 | Jouva | mmmm |
| 14:48.33 | Jouva | Well, the in-house server is on the LAN, but the VoIP provider is on the public internet obviously |
| 14:48.52 | Jouva | So we're natted between the provider and our Asterisk server, but the machine would be on the same network |
| 14:49.21 | Jouva | Err... the fax machine would be on the same network as the T.38 supporting Asterisk server. There, less ambiguous |
| 14:49.54 | carrar | You might some ATA's don't do well behind NAT with T.38 |
| 14:50.34 | carrar | But removing the whole NAT issue, T.38 works great |
| 14:50.47 | Jouva | Probably going to get the Linksys SPA device |
| 14:50.48 | hurdman | [TK]D-Fender: is there a way to post "somewhere" a backported-patch for 1.6 if it works ? |
| 14:50.56 | drmessano | What is the issue with T.38 and NAT? |
| 14:51.13 | carrar | He hasn't had issues yet |
| 14:52.03 | Jouva | Ok, but basically I won't need to care if my VoIP provider explicitly supports T.38? |
| 14:52.31 | carrar | You will if they don't negotiate T.38 and you need it |
| 14:52.50 | [TK]D-Fender | hurdman, You got a web-server? Feel free to host it.. or shove it on the tracker, but "dev" is dead |
| 14:52.54 | Jouva | Mmmm, yeah I'm using voip.ms and they said they don't have T.38 support |
| 14:53.04 | carrar | find someone who does |
| 14:53.18 | hurdman | [TK]D-Fender: i got some web-server , thx ;) |
| 14:53.31 | Jouva | We're happy with voip.ms otherwise and just put money into them and such |
| 14:53.46 | Jouva | Losing fax isn't a big issue but I just wanted to see if we could still keep the machine to make it "easy" |
| 14:54.00 | leifmadsen | flowroute is awesome for T.38 faxing |
| 14:54.22 | leifmadsen | tested recently for an update to the fax chapter in ATDG 4e, and it works quite well |
| 14:54.29 | *** join/#asterisk bchia (~Adium@nat/digium/x-gxuccxcattjvjmjd) |
| 14:54.42 | carrar | I would require T.38 from any provider |
| 14:55.08 | leifmadsen | I just use separate providers where necessary |
| 14:55.25 | leifmadsen | the 2 most popular canadian carriers (unlimitel and voip.ms) don't support t.38 |
| 14:55.39 | carrar | lame |
| 14:55.45 | carrar | THey need to get on the ball |
| 14:55.46 | leifmadsen | so not much of a choice there. Decided on flowroute just for the fax stuff since they have canadian dids |
| 14:55.51 | leifmadsen | shrugs |
| 14:55.53 | Jouva | They're looking into it I think |
| 14:55.54 | leifmadsen | neither are my company |
| 14:56.05 | carrar | They should purchase some Sonus's |
| 14:56.06 | carrar | heh |
| 14:56.24 | coppice | he swore |
| 14:56.32 | drmessano | [10:49] <carrar> You might some ATA's don't do well behind NAT with T.38 <--- I was asking in general. Is there a specific issue with T.38 behind a NAT, or the devices themselves, or....? |
| 14:56.54 | carrar | I've had issues where the * was outside and the ATA was inside |
| 14:58.15 | coppice | drmessano: when UDPTL is used it generally runs on different ports from the RTP, and the extra holes often don't get punched through the firewall. I found if you reuse the same ports things are a lot more reliable |
| 14:58.20 | carrar | But alas that is not his case |
| 14:58.21 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 14:59.08 | carrar | Just things to watch out for is all |
| 14:59.26 | carrar | Makes for GOOD TIMES |
| 14:59.27 | carrar | heh |
| 15:00.04 | *** join/#asterisk LgK (~chatzilla@94.143.84.95) |
| 15:00.09 | LgK | Hi |
| 15:02.54 | *** join/#asterisk SerajewelKS (devnull@wikipedia/Crazycomputers) |
| 15:03.00 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 15:12.43 | Jouva | carrar: If I do use T.38 with a T.38 provider for sending and receiving faxes on a phone number that's dedicated for faxing, would it just be treated like any other call? |
| 15:13.28 | Jouva | Like out going calls just tell the VoIP provider to make a call and incoming calls would Dial() the SIP line the fax is connected to? |
| 15:14.24 | raub | fail2ban question: should the rules in /etc/fail2ban/filter.d/asterisk.conf for 1.8 look like http://pastie.org/private/evg6jcpxxfihs7xagjpng ? |
| 15:15.52 | drmessano | Jouva: Yes.. |
| 15:17.15 | LgK | is there someone who can help me with the callerid function, I need to change the text when the caller phone number is hide, actually it's "asterisk" but I want to change it, but I don't found the exactly commande in the core show function callerid |
| 15:19.47 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:22.52 | *** join/#asterisk sbszulu (~dundubala@105.242.244.22) |
| 15:24.00 | ChannelZ | LgK: Set(CALLERID(name)=Fooo) |
| 15:26.36 | LgK | ok ChannelZ, so after that the "asterisk" text on the phone is replace by "Fooo" text, I juste need to replace "Fooo" with my text like "HiddenNumber" |
| 15:27.31 | ChannelZ | Yeah |
| 15:28.58 | LgK | ok |
| 15:29.01 | LgK | thanks |
| 15:36.39 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-fbhavkzmimnsduwa) |
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| 15:57.41 | Greenlight | I'm getting audio quality issues when running a large number of conferences (ConfBridge). The server has plenty of resources spare. It's running Asterisk 10.6.1. My network conditions seem fine (100mb internet pipe and the ASA shows loads spare). I recently changed from res_timing_timerfd to res_timing_dahdi which seemed to cure the issue but now I'm getting it again. How can I go about |
| 15:57.41 | Greenlight | diagnosing this? |
| 16:01.44 | Greenlight | Is a hardware timing source still recommended these days, such as a UT50 ? |
| 16:01.59 | *** join/#asterisk keycruncher (~Adium@oh-67-76-244-251.sta.embarqhsd.net) |
| 16:02.00 | shido6 | define large number of conferences as well as 'server has plenty of resources' - elaborate on 'resources' |
| 16:02.12 | shido6 | please and thank you |
| 16:04.30 | Greenlight | 150 or so conferences with 2 participants, and a confbridge recorder. Servers is Dell R510, dual Xeon E5620 2.4Ghz (16 cores(HT)), 24Gb memory. "top" reports asterisk process running at 300% - 550%. Over 10Gb free memory. |
| 16:06.08 | Greenlight | Annoyingly this has been running fine for past few weeks, but we've been gradually increasing the number of users |
| 16:06.14 | shido6 | sip? pri? |
| 16:06.20 | Greenlight | Sorry, all SIP |
| 16:06.23 | Greenlight | Alaw exclusively |
| 16:07.37 | Greenlight | Running CentOS 6.3 too |
| 16:11.30 | Greenlight | ANy ideas? Could this be server load (approx 25% of CPU capacity)? Or timing? |
| 16:12.20 | shido6 | its the recording |
| 16:12.28 | shido6 | it smells like the recording, I should say - |
| 16:12.42 | shido6 | what media are you recording to? |
| 16:12.44 | Greenlight | Recordings are done to /dev/shm |
| 16:13.01 | Greenlight | And moved as a whole to either local media or NAS |
| 16:13.23 | shido6 | when does that happen? |
| 16:13.41 | shido6 | :) |
| 16:13.51 | Greenlight | It's a bit of a hack, but we fire an Originate via the AMI, which calls system to move the call |
| 16:13.59 | Greenlight | THis happens after the call is completed |
| 16:14.15 | Greenlight | Or else /dev/shm would fill up quick (100k+ calls a day!) |
| 16:20.44 | Greenlight | So, what makes you think recording is to blame? |
| 16:20.49 | navaismo | Greenlight, /dev/shm is a ram location? im agree with shido6 i think your issue is in the recordings |
| 16:21.18 | Greenlight | Yes, /dev/shm is a ramdisk |
| 16:22.43 | navaismo | hmm |
| 16:22.47 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 16:27.10 | shido6 | tell me about your has? |
| 16:27.12 | shido6 | nas |
| 16:27.45 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:28.02 | Greenlight | It's just a server with a RAID array running Windows 2008 |
| 16:28.11 | Greenlight | I've a samba share mounted to the Asterisk box |
| 16:28.14 | Qwell | well there's your problem |
| 16:28.31 | Greenlight | Ok - will bottlenecks at this point, cause audio issues? |
| 16:28.42 | Greenlight | I do notice high queue lengths in "top" at times |
| 16:28.59 | Qwell | Show us the output from top. |
| 16:29.04 | Greenlight | Two secs |
| 16:29.21 | Greenlight | top - 18:32:42 up 13:25, 2 users, load average: 5.59, 13.12, 17.15 |
| 16:29.21 | Greenlight | Tasks: 473 total, 1 running, 472 sleeping, 0 stopped, 0 zombie |
| 16:29.21 | Greenlight | Cpu(s): 4.7%us, 16.4%sy, 0.0%ni, 78.3%id, 0.0%wa, 0.0%hi, 0.6%si, 0.0%st |
| 16:29.21 | Greenlight | Mem: 24589996k total, 15937500k used, 8652496k free, 198948k buffers |
| 16:29.22 | Greenlight | Swap: 24575992k total, 0k used, 24575992k free, 14274028k cached |
| 16:29.22 | Greenlight | PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND |
| 16:29.22 | Greenlight | 3178 asterisk -11 0 9790m 287m 10m S 313.5 1.2 1938:48 asterisk |
| 16:29.23 | Greenlight | 4 root 20 0 0 0 0 S 0.7 0.0 2:43.54 ksoftirqd/0 |
| 16:29.27 | Qwell | ... |
| 16:29.30 | Qwell | ~pastebin |
| 16:29.30 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:29.31 | Greenlight | Hmm |
| 16:29.33 | Greenlight | lol |
| 16:29.34 | Greenlight | Yea ;) |
| 16:29.46 | Greenlight | Two more secs... |
| 16:30.02 | Qwell | Press 1 first. |
| 16:30.13 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 16:30.23 | *** join/#asterisk mobile_gordita (~Robert@66-87-92-250.pools.spcsdns.net) |
| 16:30.28 | Greenlight | http://pastebin.com/NHrFnRtF |
| 16:30.49 | Greenlight | See load averages are high, but CPU isn't anywhere near full load |
| 16:30.53 | Qwell | Looks perfectly reasonable. |
| 16:31.10 | Greenlight | Asterisk can peak up to 550% |
| 16:31.31 | Greenlight | So - not the remote storage or recordings ? :( |
| 16:31.56 | Qwell | There is definitely a lot of hardware access there. |
| 16:32.14 | Qwell | Where are you storing it, exactly? |
| 16:32.19 | Greenlight | What do you mean by hardware access? |
| 16:33.44 | Qwell | top doesn't exactly say what hardware is being accessed. |
| 16:34.22 | Greenlight | Around 50% of the calls are moved from /dev/shm to /mnt/recordings which is a mounted samba share. The other 50% are stored locally on a RAID10 array, with a H700 controller and BBU. Write through is enabled. |
| 16:34.33 | Qwell | It's almost certainly caused by the recordings. |
| 16:36.07 | Greenlight | Ok, so would delays/bottlenecks in moving the recordings cause choppy audio, even if while recording happens it's all to /dev/shm? Is it because the CPUs are doing a lot of context switches to handle the queued traffic ? |
| 16:36.52 | Greenlight | AM just trying to better understand the situtation so I can formulate a solution - your wisdom is greatly appricaited |
| 16:37.02 | Qwell | Have you verified that you can actually write as much as you are, to that RAID? |
| 16:37.25 | Greenlight | The local RAID or the remote one ? |
| 16:37.40 | Qwell | Anything Asterisk is involved in writing to. |
| 16:38.01 | Greenlight | Does a system call in the dialplan with a "mv" count? |
| 16:38.23 | Qwell | maybe |
| 16:40.10 | Greenlight | The local RAID has plenty of capacity that's for sure. The remote mount, perhaps not so |
| 16:40.27 | Qwell | I didn't say capacity. |
| 16:40.40 | Greenlight | Sorry, I don't mean storage capacity |
| 16:40.52 | Greenlight | I mean bandwidth and iops capacityu |
| 16:40.55 | Qwell | Throughput and performance is what you really care about. |
| 16:42.38 | Greenlight | If I was to take the move operation away from Asterisk; for instance setup a cron job instead, do you think this would help? |
| 16:42.45 | Qwell | maybe. |
| 16:44.12 | Greenlight | Ok - gonna try playing around with how things are moved after call completed. But you reckon I was looking in the wrong place when I was thinking it was a timing source problem |
| 16:47.31 | *** join/#asterisk TSM (~the_softw@fw-lon1.wenn.com) |
| 16:48.43 | shido6 | you are playing with fun stuff, Greenlight. Find the magic $auce and you could have something there *wink* |
| 16:49.32 | Greenlight | hehe - it's great fun when it works, or even when it breaks on a dev box. Not so much fun when things go wrong in production heheh ;) |
| 16:50.46 | Qwell | Greenlight: How are you writing the recordings? What format? |
| 16:50.53 | Greenlight | wav |
| 16:50.56 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:50.56 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:51.02 | Qwell | What is the native codec on those calls? |
| 16:51.04 | Greenlight | They're converted to mp3 overnight, and the database is updated |
| 16:51.06 | Greenlight | alaw |
| 16:51.08 | Greenlight | hmm |
| 16:51.11 | Greenlight | I see your point |
| 16:51.16 | Qwell | yar, record as alaw |
| 16:51.22 | Qwell | alaw>slin is cheap, but it's still a lot |
| 16:51.43 | Greenlight | Only thing is I need to have the calls available to play on the same day |
| 16:52.04 | Greenlight | Perhaps a seperate process converting, instead of Asteris k ? |
| 16:52.14 | Qwell | sure |
| 16:53.07 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:53.21 | Greenlight | Cool - ok I'll get to it! This plan is more fun than my original plan which was to rush to the Post Office and express post a Sangoma UT50 USB timing stick I have to the customer :) |
| 16:55.01 | Qwell | Greenlight: You realize those are completely worthless, right? |
| 16:55.18 | Qwell | They provide exactly zero benefit. |
| 16:57.15 | Greenlight | Hmm great |
| 16:57.38 | Greenlight | Why they worthless? I read on some forums that they were recommended |
| 16:57.49 | Qwell | I read that the moon landing was fake. |
| 16:58.11 | WIMPy | Those were the days... |
| 16:58.21 | *** join/#asterisk cklimos (~Claude@209.5.121.227) |
| 16:58.30 | Greenlight | Ok - point taken ;) It's hard to learn new stuff like this when there is so much misinformation |
| 16:59.03 | Greenlight | So their market is purely gullable people (like me!) then ? |
| 17:01.07 | *** join/#asterisk andygraybeal (~andy.gray@obsidian.casanueva.com) |
| 17:01.33 | Greenlight | I've a TE410P not in use - would that be more use as a timing source? |
| 17:02.16 | WIMPy | You shouldn't really need any hardware today. |
| 17:02.37 | Greenlight | Ok - am totally barking up the wrong tree with my timing idea then |
| 17:03.10 | Greenlight | Just like when I am getting issues and I do "timing test" i see things "slipping" and linked the two in my head. |
| 17:03.45 | WIMPy | That OTOH should not happen. |
| 17:04.49 | Greenlight | Indeed |
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| 17:06.07 | *** join/#asterisk k610 (~Instantbi@host-78-129-3-116.brutele.be) |
| 17:06.24 | Greenlight | I'm now hoping that the recordings issues that Qwell and shido suggested are the actual cause, and they are also knocking the timing test off too |
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| 17:20.42 | TSM | any ideas why S4A (skype4asterisk) no longer is recognising the DTMF being passed though from the skype client, i can see im receiving the messages in the skype debug log but my ivr does not react |
| 17:24.02 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-akhtvexjfsyujkbz) |
| 17:29.55 | *** join/#asterisk sbszulu (~dundubala@105.242.244.22) |
| 17:33.53 | Jouva | I see that in sla.conf I can make a device template by defining the section with (!) at the end and using that section name after each device section. Can I do the same for a trunk? |
| 17:34.29 | Qwell | Jouva: That's a general config file parsing thing. It works everywhere. |
| 17:34.38 | Jouva | Ok cool |
| 17:35.12 | Jouva | Kinda thought so but wasn't entirely sure |
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| 17:52.12 | Jouva | How do I reload sla.conf ? |
| 17:58.16 | keith4 | you have to reload meetme, don't you? |
| 17:58.31 | leifmadsen | yes |
| 17:58.35 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
| 17:58.35 | leifmadsen | module reload app_meetme.so |
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| 18:58.36 | ChannelZ | TSM: hmm I just noticed my Skype stopped working completely. It's pretty much dead anyway |
| 19:00.07 | ChannelZ | actually it just looks like my module went tits-up.. lets see what my DTMF is doing |
| 19:00.18 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 19:00.29 | *** join/#asterisk anonymouz666 (~anonymouz@186-241-67-55.user.veloxzone.com.br) |
| 19:01.41 | ChannelZ | Hrrm. How do you even get the dialpad up under the newer versions? |
| 19:02.30 | leifmadsen | dialplan up? |
| 19:02.36 | leifmadsen | dialplan show ? |
| 19:02.45 | leifmadsen | totally misread that |
| 19:02.46 | leifmadsen | dialplan* |
| 19:02.51 | ChannelZ | sorry, I mean the dialpad on the Skype client |
| 19:02.52 | leifmadsen | errr... dialpad |
| 19:02.58 | leifmadsen | ya, I figured it out after reading again :) |
| 19:03.02 | ChannelZ | there used to either be a button or it was just there (I don't even remember) but now I have no idea |
| 19:03.06 | leifmadsen | my bad |
| 19:03.11 | ChannelZ | no worries |
| 19:05.47 | ChannelZ | ah finally found it. My DTMF works |
| 19:06.59 | *** join/#asterisk _Corey_ (~chatzilla@mobile-166-147-114-227.mycingular.net) |
| 19:18.36 | *** join/#asterisk fireman_biff (~biff@65.48.222.203) |
| 19:19.21 | fireman_biff | Hi, is it true that dahdi always uses hardware echo cancellation when its available, even if system.conf says to use oslec? |
| 19:20.17 | jpsharp | No, you can disable the hardware stuff and use software. |
| 19:22.06 | fireman_biff | so i would need to change "echocanceller=oslec" etc... to "echocanceller=hwec"etc... right? |
| 19:22.15 | fireman_biff | if i wanted to use hardware |
| 19:22.41 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:26.36 | jpsharp | I believe so, yes. |
| 19:27.12 | fireman_biff | cool, thanks |
| 19:31.11 | jmls | is anyone aware of any problems with cisco 79xx and hold ? |
| 19:31.22 | leifmadsen | stopped reading at 79xx |
| 19:31.35 | jmls | leifmadsen: oic ;) |
| 19:31.59 | leifmadsen | jmls: I was thinking there was an option in sip.conf, but just remembered that was for mwi |
| 19:32.04 | jmls | No moh is played, and often takes 3-4 attempts to get the call off hold |
| 19:32.24 | jmls | asterisk 11 beta2 / rc1 / not sure ;) |
| 19:32.38 | jmls | other phones apparantly work ok ** |
| 19:32.50 | jmls | ** not verified by anyone trustworthy like me ... |
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| 19:35.48 | *** join/#asterisk _Corey__ (~chatzilla@mobile-166-147-113-028.mycingular.net) |
| 19:39.15 | Greenlight | Hi again - I was discussing earlier an audio quality issue I've been having. It was suggested that the problem was the recordings I was making, and perhaps a disk bottleneck. I've now altered it so they are just staying in ram (/dev/shm) and the issue still persists. |
| 19:39.51 | Greenlight | I'd previously thought it could be a timing related issue |
| 19:40.17 | leifmadsen | I don't think timing interfaces are used for recording files |
| 19:40.20 | Greenlight | The call quality on my ConfBridge conferences is terrible |
| 19:40.35 | leifmadsen | using res_timing_dahdi? |
| 19:40.39 | Greenlight | I am now |
| 19:40.55 | Greenlight | Changed from timerfd -> dahdi as per your suggestion yesterday |
| 19:41.02 | Greenlight | Seemed to fix it till about 1pm today |
| 19:41.10 | leifmadsen | I'm out of ideas. Haven't run into that issue in... years |
| 19:41.17 | Greenlight | That sucks ;( |
| 19:42.09 | Greenlight | Even under load, do you get decent results from both "timing test" and "dahdi_test" ? |
| 19:42.22 | Greenlight | I'm trying to work out if that's my problem or if it's something else |
| 19:42.41 | Greenlight | I also have rather high load average on the server - but it's been like that for weeks, without audio problems |
| 19:47.46 | jpsharp | Is there a console command to see what timing source asterisk is using? |
| 19:48.02 | Greenlight | module show like timing |
| 19:48.15 | Greenlight | You should see all timing sources and a usage count |
| 19:48.32 | jpsharp | timing test |
| 19:48.37 | Greenlight | That too |
| 19:48.58 | jpsharp | Ah, timerfd |
| 19:49.35 | Greenlight | Could hyper-threading potentially cause issues with ConfBridge under load - anyone any experience with that? |
| 19:51.40 | Greenlight | The more I Google it, the more Hyper Threading sounds like a potential cause |
| 19:53.53 | *** part/#asterisk fireman_biff (~biff@65.48.222.203) |
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| 19:55.45 | jpsharp | Greenlight: Yes, I've had issues with HT and Asterisk. |
| 19:56.41 | Greenlight | Amazing, it does seem to make sense |
| 19:56.52 | Greenlight | Seesh how didn't I think of that before. |
| 19:57.02 | Greenlight | Out of interest what issues did you have ? |
| 19:57.08 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 19:58.01 | jpsharp | grainy audio on a pair of PRIs. |
| 19:58.18 | Greenlight | And turning off HT sorted it right out |
| 19:58.25 | jpsharp | Yep. |
| 19:58.29 | Greenlight | Perfect! |
| 19:58.39 | Greenlight | I really hope that it does the same for me |
| 19:58.43 | jpsharp | No guarantee that it will fix your problem. :) |
| 19:58.57 | Greenlight | I can but hope |
| 19:59.15 | jpsharp | Since then, I've turned HT off on every machine I use. |
| 19:59.29 | jpsharp | HT is a bad hack anyway. |
| 19:59.44 | Greenlight | hmm.. is turning off HT one of these things everyone else does on their Asterisk boxes, and I've just missed the memo on ? :) |
| 19:59.58 | *** join/#asterisk talonawd91 (~Jeff@168-93-163-9.ipv4.firstcomm.com) |
| 20:02.04 | ChannelZ | I've never had it on any box, asterisk or not, ever. |
| 20:03.26 | Greenlight | It's on by default on most boxes I've deployed recently, and I always reckoned that the hardware HT must be somehow more effcieint than doing a a normal context switch, so left it on. Silly me |
| 20:04.00 | Greenlight | Now, to hope there is some onsite IT to play with the BIOS.... |
| 20:04.56 | dexteruk | Has anyone used a2billing? When i set the CALLERID to anonymous in the FROM header i see asterisk@domain |
| 20:05.06 | dexteruk | instead of anonymous@domain |
| 20:05.52 | dexteruk | I know its a2billing as i have removed it from the dialplan and routed the call out directly and the From Header is anonymous@domain |
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| 20:26.10 | Amnesia | hi, question, I've been looking around a bit |
| 20:26.37 | Amnesia | but can't seem to find an alternative to h323 for video conferencing |
| 20:26.50 | Amnesia | does one of you folks by any chance know what I;m looking for? |
| 20:27.31 | *** join/#asterisk sawgood (~sawgood@69.199.155.13) |
| 20:27.31 | Amnesia | I'm looking for some multi platform supported protocol, which is encrypted, and easy to set up (video + text only) |
| 20:28.02 | Amnesia | and add peer2peer to that:) |
| 20:28.18 | Amnesia | am I looking for something that doesn't exist> |
| 20:29.10 | jpsharp | The only SIP video conferencing using I've used is made by Tandberg, but I don't know if it does encrypted transport. |
| 20:29.13 | file | Skype? wait, no. |
| 20:29.26 | jpsharp | nukes file from orbit. |
| 20:29.29 | Amnesia | tandberg uses h323 right? |
| 20:29.46 | jpsharp | The newer boxes use SIP. |
| 20:29.49 | jpsharp | or H323. |
| 20:29.50 | Amnesia | I've tried 2 ekiga clients, but it sucks |
| 20:30.12 | Amnesia | ah so there're only propiertary solutions ? |
| 20:30.30 | Amnesia | (SIP is only a control channel right? ) |
| 20:30.58 | jpsharp | used standard SIP and RTP. |
| 20:31.25 | jpsharp | I passed Tandberg audio through Asterisk, that was before Asterisk got video support. |
| 20:32.57 | jpsharp | I also used H323 on both the Tandberg and Asterisk, but that was about as much fun as being flailed with spaghetti. |
| 20:33.25 | Amnesia | hm, so RTP can be video or sound right? |
| 20:34.19 | jpsharp | Yes. |
| 20:34.41 | TriJetScud | hey, you guys got any did's and term services at iCall? |
| 20:35.16 | Amnesia | jpsharp: so what'd be the easiest solution for a one on one videoconference between tux and m$ ? |
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| 20:38.42 | Amnesia | O |
| 20:38.52 | Amnesia | I'm fed up with fscking msn |
| 20:46.04 | ChannelZ | maybe just kiss them a little first. |
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