| 00:05.52 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 00:14.01 | *** join/#asterisk Mango45 (~Mango45@209.89.211.69) |
| 00:17.11 | Mango45 | How do you configure sip.conf to authenticate a phone by IP? I put host=192.168.10.5 in, but it allows any phone with that username to make calls as that peer, regardless of what IP address it has. |
| 00:21.29 | Mango45 | My sip.conf: http://pastebin.ca/2429300 |
| 00:34.29 | Mango45 | hah. type=peer |
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| 01:09.52 | *** part/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
| 01:14.12 | zendel | hi, I've pasted a Query in http://pastebin.com/PSABQfcn. The question is regarding "queue member ackcall - cpuspikes". Can anybody give some input ? |
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| 01:25.04 | navaismo | ~upgrade |
| 01:25.04 | infobot | Upgrading is easy! Go that way, really fast. If something gets in your way, turn. |
| 01:25.33 | navaismo | maybe on recent version asterisk dont do that |
| 01:26.27 | zendel | <navisimo>: This is in response to my Query ? Upgrading is not an option. |
| 01:27.20 | zendel | Cos there's are other APPS running. |
| 01:29.04 | navaismo | hmm too bad, but what apps are you using that doesn't in recent versions of asterisk? |
| 01:30.45 | zendel | its about the AELs , macros & there are lot of them. Changing asterisk version means rewriting stuff which does not work. And it's a risk for a production system. Is this a bug OR it has to do with the Dial Plan ? |
| 01:30.49 | floren_ | zendel: why upgrading is not an option |
| 01:31.07 | floren_ | latest version of asterisk has tons of features and enhancements |
| 01:33.02 | zendel | This will require major re-coding. Cos AEL/Macros has evolved. |
| 01:33.11 | floren | i see |
| 01:33.19 | floren | what versionn you run now? |
| 01:33.36 | zendel | 1.6.1.20 |
| 01:34.16 | floren | hmm ChannelZ or [TK]D-Fender might be able to help, i'm working my way up on knowledge with 11.x only |
| 01:35.05 | zendel | ok. |
| 01:35.19 | zendel | thx |
| 01:35.25 | *** join/#asterisk navaismo (~navaismo@189.241.78.204) |
| 01:36.34 | [TK]D-Fender | that isn't even the latest from that branch. |
| 01:37.07 | [TK]D-Fender | and neither it, nor the one one that followed is supported at all any further |
| 01:37.28 | [TK]D-Fender | locking youself to it is suicide |
| 01:37.43 | floren | heh [TK]D-Fender |
| 01:38.02 | floren | [TK]D-Fender: how are you |
| 01:38.13 | [TK]D-Fender | shitty... but alive... |
| 01:38.17 | floren | :) |
| 01:38.33 | zendel | Well i know, upgrading now is asking for trouble. All I need to know is whether it's a BUG or some crappy dial plan. |
| 01:38.41 | floren | you're not going to like it, about to set a 2nd crisco 8961 in my appartment :D |
| 01:38.47 | [TK]D-Fender | I got my shirt off today :) |
| 01:38.48 | zendel | if it's a BUG i'll leave it alone |
| 01:38.56 | floren | heh [TK]D-Fender |
| 01:39.47 | [TK]D-Fender | zendel: cpu spike isn't some little thing you should be doing in th dialplan .... it's the apps |
| 01:40.01 | [TK]D-Fender | and the rest of the core |
| 01:40.12 | zendel | ok |
| 01:41.10 | [TK]D-Fender | which means it isn't going to get better... |
| 01:41.27 | floren | zendel: upgrade time :) |
| 01:41.59 | [TK]D-Fender | I'd start the process of digging your way out of obsolescence sooner than later.... |
| 01:42.42 | [TK]D-Fender | I've seen several dumb projects screw themselves locking to 1.4... |
| 01:43.41 | [TK]D-Fender | good at the time tends to be found to be worse over time |
| 01:44.25 | navaismo | so i'm getting this with Blink, Jitsi, Phoner & MicroSIP--> == Problem setting up ssl connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca |
| 01:44.25 | navaismo | [Aug 8 01:43:12] WARNING[29105]: tcptls.c:261 handle_tcptls_connection: FILE * open failed! |
| 01:48.48 | *** join/#asterisk thebmw (~thebmw@i.am.thebmw.net) |
| 01:50.53 | leifmadsen | zendel: sounds like a bug, but without looking at the dialplan or output of anything, it'll be hard to confirm anything |
| 01:51.01 | leifmadsen | but a cpu spike sounds like a bug to me |
| 01:51.11 | leifmadsen | 1.6.1.20 is pretty old, and hasn't been supported for at least a couple of years |
| 01:51.48 | *** part/#asterisk IAmTheClaw (~IAmTheCla@65-128-185-96.mpls.qwest.net) |
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| 02:41.37 | asteriskmonkey | heya heres a wierd one, |
| 02:41.51 | asteriskmonkey | there is gotoiftime functions in asterisk |
| 02:42.04 | asteriskmonkey | but i dont seem to see anythhing to noop the current time as the pbx sees it |
| 02:42.22 | asteriskmonkey | is there a command for this? |
| 02:43.15 | asteriskmonkey | ${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)}) |
| 02:43.17 | asteriskmonkey | found it nm |
| 02:44.06 | navaismo | man this tls/srtp stuff sucks neither Blink/Bria/Jitsi/Phoner/MicroSip/SFLphone/Linphone work on it |
| 02:45.16 | *** join/#asterisk mintos (~mvaliyav@1.186.102.211) |
| 02:47.15 | [TK]D-Fender | or you just did it wrong |
| 02:47.28 | [TK]D-Fender | places bets |
| 02:48.48 | navaismo | well blink/bria/MicroSIP cant login because the error---> == Problem setting up ssl connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca [Aug 8 01:43:12] WARNING[29105]: tcptls.c:261 handle_tcptls_connection: FILE * open failed! |
| 02:50.06 | navaismo | Phoner/Jitsi doesn't support srtp diling show the error---> [Aug 8 02:47:39] WARNING[29849][C-0000001c]: chan_sip.c:10469 process_sdp: Matched device setup to use SRTP, but request was not! |
| 02:50.21 | [TK]D-Fender | zopen failed = don't trust file or contents |
| 02:50.56 | [TK]D-Fender | I trust nothing obviously |
| 02:51.09 | navaismo | and i followed step by step the recipe from asterisk wiki blink didn't work. |
| 02:51.46 | navaismo | then my T20P from yealink cant send the invite correctly if SRTP is enable if I disbale SRTP all work great using tls |
| 02:51.56 | [TK]D-Fender | stop wallowing and show all the backup] |
| 02:52.13 | [TK]D-Fender | "I followed the guide" = WEAK SAUCE |
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| 02:56.15 | navaismo | spent all dat between certificates, vpn settings and cap traces grrr |
| 03:02.12 | asteriskmonkey | heres a wierd one, im using asterisk in a vm environment |
| 03:02.22 | asteriskmonkey | local time is set correctly however asterisk is using utc |
| 03:02.36 | asteriskmonkey | where do i change it from utc? |
| 03:06.36 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 03:07.35 | asteriskmonkey | im using a gotoiftime and its driving me nuts on this vm box because asterisk keeps using utc |
| 03:10.04 | navaismo | http://www.centos.org/docs/5/html/5.1/Deployment_Guide/s2-sysconfig-clock.html |
| 03:16.25 | asteriskmonkey | thanks was missing a timezone file :P |
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| 04:01.13 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 06:08.23 | igcewieling | 2 hours of debugging a freepbx upgrade and apparently all I had to do was log out of the GUI and log back into it. |
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| 06:31.19 | ChannelZ | isn't FPBX grand? |
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| 07:43.51 | mnathani | how can I setup a webpage where I can enter 2 phone numbers and have asterisk dial both and conference them together? |
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| 07:47.39 | ChannelZ | call files |
| 07:48.04 | ChannelZ | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files |
| 07:49.01 | mnathani | ChannelZ: Thanks |
| 07:50.13 | ChannelZ | Yup. Simple matter of making your website generate call files and dump them in a ConfBridge, or into your dialplan that ultimately puts them into a conference if you need to do other things first |
| 07:51.15 | ChannelZ | although that assumes your * box is the same as the web server. But you can come up with some means to do it if they are different servers |
| 07:53.43 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:64ae:f04f:b795:e1af) |
| 07:53.45 | ChannelZ | could also do it via sockets if they are remote, via Manager. |
| 07:54.16 | ChannelZ | More than one way to skin a cat. And with that I'm off to bed. |
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| 08:04.19 | *** join/#asterisk bombev (~User@PPPoE-Static-40-132.UnicsBG.Net) |
| 08:04.26 | bombev | Hi guys |
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| 08:07.07 | bombev | I was wondering is it possible to be created some dialplan which will call bunch of extension and then to play some announcenement with BackGround and then clients to be able to choose what to happens |
| 08:08.03 | WIMPy | sure |
| 08:09.13 | bombev | How should I start WIMPy can you give me some advice |
| 08:10.03 | WIMPy | Did you read the book? |
| 08:10.19 | bombev | yes |
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| 08:10.45 | WIMPy | Then you should know what you need. |
| 08:11.02 | WIMPy | Some way to generate a call and the rest is just the normal IVR stuff. |
| 08:11.17 | bombev | got it |
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| 08:44.52 | uyulala | hi all |
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| 09:03.16 | WIMPy | Yay! Finally a pickup from VM. |
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| 09:14.34 | WIMPy | still wonders how to sensibly do it for multiple users, but for now I'm quite happy. |
| 09:14.44 | peetaur | Is there a sound quality tuning guide somewhere? Even on the local gigabit LAN, sometimes there is jitter and bits of silence where there should be talking. It's much more common with the recordings than with 2 people talking. |
| 09:14.54 | peetaur | (and it's just a test so far ... no heavy traffic) |
| 09:15.38 | WIMPy | A bad timing source? Try 'timing test'. |
| 09:16.55 | peetaur | where is timing test? |
| 09:17.02 | WIMPy | On *CLI |
| 09:17.34 | peetaur | It has been 1000 milliseconds, and we got 50 timer ticks |
| 09:17.43 | peetaur | is that good? :D seemed quick and simple for a test\ |
| 09:18.01 | WIMPy | Yes and yes. |
| 09:18.10 | WIMPy | Or at least not obviousely bad. |
| 09:18.42 | WIMPy | Which timing module do you use? |
| 09:19.24 | peetaur | how do I find out? I'm using asterisk 11 + FreePBX 2.11-rc1 |
| 09:19.42 | WIMPy | It told you in the first of the two lines of output. |
| 09:19.54 | peetaur | ah, I see Using the 'timerfd' timing module for this test. |
| 09:20.02 | WIMPy | ok |
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| 09:21.07 | tompaw | Hey, great job with 11.5! Why the f*ck would we need rtp engine installed by default anyway? |
| 09:22.12 | WIMPy | Only works if you have the neccessary dependencies installed. |
| 11:07.22 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 11:07.22 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.5.0 (2013/07/15), 10.12.2 (2013/03/27), 1.8.23.0 (2013/07/15), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 11:35.27 | phix | Greenlight: yes, that is bad |
| 11:35.31 | phix | m'kay |
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| 12:43.15 | level7 | they do what? |
| 12:43.21 | level7 | Greenlight: what do you mean? |
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| 13:40.43 | abradley | I'm interested in setting up some desk phones (ip phones) at a couple of cust serv reps' homes for instances such as bad weather. Unfortunately, we must be hipaa compliant. Can someone point me in the right direction as far as setting up polycom phones (ip450, for example) to work over ssl or vpn connection? |
| 13:41.21 | abradley | When I go into the menu on the phones I'm unable to locate any SSL or encryption settings |
| 13:41.48 | [TK]D-Fender | TLS+SRTP |
| 13:42.57 | abradley | [TK]D-Fender, thanks, I'll look into this. |
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| 13:54.42 | saxa_ | Hi, i set up asterisk 1.8.23.0 with tdm400 analog card, have 3 analog lines and sip phones. The problem is that 90% of the times when i call the line gives a busy signal. If I try to call by mobile phone to that number the number 90% of the times is not busy. |
| 13:54.56 | saxa_ | I would like to know how can I debug this issue ? |
| 13:55.09 | saxa_ | how can I understand what is wrong ? |
| 13:55.53 | saxa_ | if I go to the server disconnect the line and reconnect it it calls immediately |
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| 13:56.18 | saxa_ | for the sip phone I use grandstream phones |
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| 13:56.54 | saxa_ | can be a sip misconfiguration or is chan_dahdi problem ? |
| 13:57.05 | twitchnln | morning, anyone ever had a problem with remote extensions not establishing audio path on external calls but inter-extension calls are fine? |
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| 13:57.47 | Greenlight | twitchnln: Sounds like a NAT related issue |
| 13:57.58 | Greenlight | Do you have directmedia enabled ? |
| 13:59.11 | twitchnln | reinvites are disabled if that is what you mean |
| 14:00.13 | twitchnln | remote phone should proxy thru pbx to get to pstn, not reinvite directly to provider |
| 14:00.31 | Greenlight | And you're 100% sure? Can we see sip.conf ? |
| 14:00.45 | Greenlight | Also, what version of asterisk is this? |
| 14:01.09 | twitchnln | which portion? or all of it? |
| 14:01.17 | twitchnln | 1.4.22 |
| 14:01.42 | Greenlight | That's old.... |
| 14:01.52 | [TK]D-Fender | decrepit... |
| 14:01.57 | Greenlight | And no longer supported |
| 14:02.21 | twitchnln | it's alot better than the aah box I replaced |
| 14:02.35 | [TK]D-Fender | and when was that? |
| 14:03.15 | twitchnln | like 2 years ago |
| 14:03.20 | Greenlight | Why 1.4 then ? |
| 14:03.26 | Greenlight | 1.4 was old even then |
| 14:03.42 | Greenlight | or is this some trixbox type packed up thing? |
| 14:03.45 | [TK]D-Fender | because it was on some iso that was old when he looked then too |
| 14:04.04 | *** join/#asterisk italorossi (~italoross@67.201.69.130) |
| 14:04.05 | twitchnln | it's stable, and I have 300+ in the field, and I don't want to roll a new installer image |
| 14:04.17 | Greenlight | Ok, well good luck with your problem. |
| 14:04.31 | *** part/#asterisk twitchnln (~twitch@adsl-72-152-21-140.asm.bellsouth.net) |
| 14:04.34 | [TK]D-Fender | lets nt jum tthat far.... |
| 14:04.45 | [TK]D-Fender | just fix your nat settings |
| 14:05.28 | [TK]D-Fender | 'he had a configuration problem, not a "bug". Don't be an ass. |
| 14:05.48 | Greenlight | external users can call internal users okay, but not call externally. |
| 14:05.54 | Greenlight | And he says directmedia is disabled |
| 14:06.03 | Greenlight | Or, "reinvite" as it was back then. |
| 14:06.14 | [TK]D-Fender | that word doesn't even exist in 1.4 |
| 14:06.33 | [TK]D-Fender | and you think that is all there is to it? |
| 14:06.40 | [TK]D-Fender | it isn't |
| 14:07.04 | Greenlight | If there's no reinvites then nat must be okay for the RTP to flow for an external -> internal call |
| 14:07.39 | Greenlight | And I'm not trying to be an ass, I just don't remember stuff like the reinvite options etc for that version |
| 14:07.57 | Greenlight | Or if there were any "gotchas" |
| 14:08.33 | [TK]D-Fender | WGLWAT is a a GTFO phrase mostly... |
| 14:08.45 | [TK]D-Fender | no need to abandon him on this |
| 14:09.36 | Greenlight | Well lets see whats in sip.conf |
| 14:09.43 | Greenlight | Think he's justm getting it |
| 14:10.20 | *** join/#asterisk felipealmeida (~user@177.98.67.233) |
| 14:11.44 | Greenlight | Oh, just noticed he left |
| 14:13.00 | *** join/#asterisk felipealmeida (~user@177.98.67.233) |
| 14:20.30 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 14:23.55 | *** join/#asterisk boom^time (~boom^time@75-151-20-174-Michigan.hfc.comcastbusiness.net) |
| 14:24.29 | saxa_ | Hi, i set up asterisk 1.8.23.0 with tdm400 analog card, have 3 analog lines and sip phones. The problem is that 90% of the times when i call the line gives a busy signal. If I try to call by mobile phone to that number the number 90% of the times is not busy. |
| 14:24.32 | level7 | so anybody used kolmisoft MOR and can provide me some feedback? |
| 14:24.43 | level7 | or if you have tested some other billing solution, let me know :) |
| 14:24.58 | *** join/#asterisk SuPrSluG (~SuPrSluG@rrcs-50-75-185-122.nys.biz.rr.com) |
| 14:25.05 | boom^time | Hey guys. I haven't used asterisk since <1.4 days just got back into it. Trying to limit my rtp port range down to just 1000 ports for experimentation. However, if I modify rtp.conf asterisk 11.5 doesn't seem to obey. Has something changed? |
| 14:25.25 | duchman | <level7>have used a2billing for about 2yrs..... |
| 14:25.42 | level7 | :) |
| 14:25.50 | Greenlight | boom^time: Did you restart it after making the changes ? |
| 14:25.57 | boom^time | Yes :) |
| 14:26.07 | level7 | well I know that who is using a2billing is quite sadisfied of the thing |
| 14:26.20 | duchman | seems ok when u get past the many complex parts |
| 14:26.28 | duchman | esp rates thing |
| 14:26.43 | [TK]D-Fender | boom^time: and you retarted *? |
| 14:26.57 | boom^time | Yes :) |
| 14:27.03 | Greenlight | Dejavu |
| 14:27.28 | [TK]D-Fender | boom^time: show us the configs and a call with sip debug enabled |
| 14:34.43 | boom^time | Okay http://pastebin.com/q6JvJPJk |
| 14:35.36 | boom^time | But I'll see this > 0x7f8b50021740 -- Probation passed - setting RTP source address to 10.1.10.70:16646 |
| 14:35.54 | boom^time | sorry forgot my rtp.conf, one sec |
| 14:35.55 | Greenlight | Isnt that the IP of your endpoint? |
| 14:36.05 | Greenlight | Or is that Asteris k? |
| 14:36.19 | boom^time | Yes that's for my sip phone |
| 14:36.32 | Greenlight | So the phone decided it's RTP port... |
| 14:36.45 | boom^time | Sure did.. hmm. |
| 14:37.15 | Greenlight | You can only control your own RTP port, not the other side |
| 14:39.21 | boom^time | Makes sense. Thanks for clearing that up for me. |
| 14:40.31 | *** join/#asterisk Rahail (~Rahail@67.214.121.181) |
| 14:40.40 | Rahail | HI there I am having ussing with asterisk u limit |
| 14:40.51 | Rahail | even i did ulimit -n 65536 |
| 14:41.05 | Greenlight | WHich user is asterisk running as ? |
| 14:41.07 | Rahail | still i get this to much file open cant modified i also change under /etc/security |
| 14:42.09 | Rahail | 1.8.120 |
| 14:42.56 | Rahail | sorry Greenlight Asterisk 1.8.21.0 built by root |
| 14:43.07 | Greenlight | Yes, and which user is it running as |
| 14:43.21 | Greenlight | Because you need to set your limits for that particular user |
| 14:43.33 | Rahail | i did ulimit -n 65536 |
| 14:43.40 | Greenlight | As root ? |
| 14:43.44 | Rahail | yes |
| 14:43.51 | Greenlight | Ok, and who is asterisk running as? |
| 14:43.56 | Rahail | root |
| 14:44.18 | Greenlight | WHat's the actual error you're getting ? |
| 14:44.34 | Rahail | [Aug 8 07:03:28] WARNING[5875]: tcptls.c:290 ast_tcptls_server_root: Accept failed: Too many open files |
| 14:45.52 | Greenlight | Do you have any limits set in asterisk.conf |
| 14:46.33 | Rahail | nope everyting is comment out |
| 14:46.36 | Rahail | udner asterisk.conf |
| 14:46.45 | Greenlight | Try uncommenting and setting there |
| 14:46.56 | Greenlight | And see if asterisk complains when starting or not |
| 14:47.05 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-113-31.user.veloxzone.com.br) |
| 14:47.16 | Rahail | ;maxfiles = 1000 |
| 14:47.22 | Rahail | is this the one i should uncomment |
| 14:47.30 | Greenlight | Yea, set that to something a lot higher |
| 14:47.44 | Rahail | ok what about |
| 14:47.53 | Rahail | . /etc/security/limit.conf |
| 14:48.12 | *** join/#asterisk slidesinger (~slidesing@c-69-141-78-33.hsd1.nj.comcast.net) |
| 14:48.17 | *** join/#asterisk MezzFA0 (~waynemerr@mail.thevoiceasia.com) |
| 14:48.21 | Greenlight | I thought you'd already increase it in there? |
| 14:48.34 | Rahail | i put name udner astierks |
| 14:48.38 | Rahail | so i should change to root |
| 14:48.39 | Rahail | right |
| 14:48.42 | Greenlight | Well |
| 14:48.48 | Rahail | asterisk soft nofile 65535 |
| 14:48.48 | Rahail | asterisk hard nofile 65535 |
| 14:48.49 | Greenlight | You really shouldn't run asterisk as root |
| 14:49.02 | file | vibrates |
| 14:49.20 | file | Fun fact: Every time "file" is used in a sentence my cellphone vibrates |
| 14:49.29 | Rahail | lol |
| 14:49.35 | Greenlight | ^^ |
| 14:49.46 | Rahail | Greenlight so i should change the asteirsk to root |
| 14:49.50 | Rahail | under limit.conf asterisk soft nofile 65535 |
| 14:49.50 | Rahail | asterisk hard nofile 65535 |
| 14:50.14 | Greenlight | Well why do you have an "asterisk" user, if you're running asterisk as root |
| 14:50.31 | Rahail | i thought it was runing as asterisk |
| 14:50.36 | Greenlight | Right , okay |
| 14:50.36 | Rahail | but now i relize its runing as root |
| 14:50.50 | Greenlight | Add entries for root in there as well. |
| 14:50.52 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:50.52 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:51.27 | MezzFA0 | hi all, I have a bit of a noob question about dialplans. Anyone know much about what context originate uses to connect two extensions? |
| 14:51.34 | *** join/#asterisk slidesinger (~slidesing@c-69-141-78-33.hsd1.nj.comcast.net) |
| 14:51.43 | Greenlight | MezzFA0: Whichever one you tell it to. |
| 14:52.40 | MezzFA0 | ok, I have an auto answer context and it works fine on my two test extensions (5001 & 5002). But when I call originate from the cli to connect 5001 to 5002, 5001 rings, I answer and then 5002 auto answers. Can I stop that first ring? |
| 14:52.40 | boom^time | So.. I'm proxying all of my traffic through a VPS. I can make outbound, but only get one way audio. The firewall my local asterisk server is behind I opened up 1000 ports both udp/tcp and set that range in asterisk. It's gotta be something to do with the fw |
| 14:52.55 | Rahail | Greenlight after teh changes |
| 14:52.58 | Rahail | do i need to reboot |
| 14:53.05 | Rahail | <PROTECTED> |
| 14:53.59 | Greenlight | Rahail: Hmm I'm not 100% sure. Try restarting asterisk and it should say something about setting max open files |
| 14:54.05 | Greenlight | See if it complains it cant |
| 14:54.07 | MezzFA0 | e.g. #>originate SIP/5001 extension 5002@my-custom-auto-answer, 5001 rings first, 5002 answers automatically |
| 14:54.25 | Rahail | where whill i see this msg |
| 14:54.41 | Rahail | as i am starting the asterisk from rc.local |
| 14:54.44 | Greenlight | MezzFA0: You likely need to use a Local channel |
| 14:54.50 | *** join/#asterisk brian98 (~brian98@unaffiliated/brian98) |
| 14:55.16 | Greenlight | Rahail: You can start it from command line using "asterisk -c" |
| 14:55.29 | MezzFA0 | Thats where I'm confused, I don't see a channel being created until I answer so I'm not sure how I know what channel to use |
| 14:55.33 | Rahail | while its running i still can do the -c |
| 14:55.53 | Greenlight | originate Local/5001@my-custom-auto-answer extension 5002@my-custom-auto-answer |
| 14:55.55 | Greenlight | Try that ^^ |
| 14:56.11 | Greenlight | Rahail: No, you'll need to stop it. |
| 14:56.27 | *** join/#asterisk vlad_starkov (~vlad_star@188.123.241.16) |
| 14:56.33 | *** join/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
| 14:56.44 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:56.48 | Greenlight | It may even try and set it just doing "asterisk -r" |
| 14:56.57 | *** part/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
| 14:57.04 | Greenlight | You should see a message about "setting max open files"... |
| 14:57.22 | boom^time | is there anyway to view rtp information? such as if it's receiving a stream and from where? |
| 14:57.26 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:57.26 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:57.48 | [TK]D-Fender | MezzFA0: there is a channel created to call the device you specified ... and when it answersds, it is dumped into the dialplan |
| 14:57.55 | MezzFA0 | Thanks Greenlight, I've wasted half the morning fiddling with that |
| 14:58.03 | MezzFA0 | Local/5001 etc works fine |
| 14:58.06 | Greenlight | :) |
| 14:58.09 | [TK]D-Fender | MezzFA0: it does not inherently "connect 2 devices" |
| 14:58.11 | *** join/#asterisk ghost75 (~trechber@dslb-188-105-016-029.pools.arcor-ip.net) |
| 14:58.27 | Greenlight | The Local will send you into the dialplan, rather than going direct to the endpoint |
| 14:58.55 | Greenlight | Presumabley setting your SIP auto-answer header |
| 14:59.26 | [TK]D-Fender | if that's what is being set... |
| 14:59.40 | Greenlight | Just guessing from the context name :) |
| 15:01.34 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:01.42 | MezzFA0 | thats correct, I managed to get as far as the SIPHeaders but the last bit completely threw me |
| 15:01.51 | ghost75 | registerattempts=0 <- this should do infinite attempts right? |
| 15:01.59 | Greenlight | Yes |
| 15:02.18 | ghost75 | hmm i had one peer not registered until i restarted * |
| 15:02.34 | ghost75 | and parameter was set to 0 |
| 15:03.05 | ghost75 | could this be problem with nat table? |
| 15:04.21 | Greenlight | That parameter is when asterisk is registering. |
| 15:04.35 | Greenlight | This sounds more like your peer was registering with asterisk. |
| 15:05.12 | Katty | hello my asterisk does not work at all how ot fix plz |
| 15:05.35 | ghost75 | last night at 2am i had mail that one peer was not registered anymore, i guess that happened during ip change of isp |
| 15:05.47 | ghost75 | since then didnt work until i restarted * |
| 15:05.56 | Greenlight | ghost75: Yea things like that are going to cause issues ^^ |
| 15:06.12 | Katty | carrar: how was your walk last night |
| 15:06.17 | Katty | carrar: with the misses.. |
| 15:06.37 | ghost75 | openwrt has kind of problem with nat table |
| 15:07.20 | *** join/#asterisk boom^time (~boom^time@75-151-20-174-Michigan.hfc.comcastbusiness.net) |
| 15:08.45 | boom^time | If you're using two NICs with your asterisk server will it accept traffic from each network or does it bind to a specific one? |
| 15:10.58 | wdoekes | boom^time: each |
| 15:11.18 | boom^time | Thank you |
| 15:11.24 | *** join/#asterisk aruntomar (~Thunderbi@49.248.158.32) |
| 15:11.34 | wdoekes | you could force it to listen to a single one if you wanted, I think |
| 15:11.39 | wdoekes | but it defaults to any |
| 15:11.45 | ghost75 | Greenlight: https://dev.openwrt.org/ticket/10225 |
| 15:11.50 | boom^time | using externip? |
| 15:11.56 | wdoekes | udpbindaddr |
| 15:12.33 | boom^time | under each sip friend/peer? |
| 15:12.41 | wdoekes | no |
| 15:12.47 | boom^time | or probably just under default |
| 15:12.57 | wdoekes | [general] |
| 15:14.16 | boom^time | haha that made this sexy error "chan_sip.c:4341 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data" |
| 15:14.48 | Greenlight | You perhaps need to specify a port as well |
| 15:15.13 | boom^time | I did, 5060. I already had bindaddr=0.0.0.0 and I changed that to my WAN ip as well |
| 15:15.26 | boom^time | also tcpbindaddr=0.0.0.0 was there and I changed that. |
| 15:15.35 | Greenlight | And that's the IP assigned to that NIC |
| 15:15.56 | Greenlight | There's no NAT ? |
| 15:15.57 | boom^time | Shoot, no. Was using my WAN IP. |
| 15:16.03 | boom^time | that should fix it brb |
| 15:17.07 | boom^time | Well it fixed it but I'm still back to my one way rtp stream. damn. |
| 15:19.43 | boom^time | not my firewall just tested random udp ports with netcat. Must be the way I'm proxying the calls |
| 15:20.02 | Greenlight | proxying the calls ? |
| 15:20.59 | boom^time | Yeah, I have this irritating setup right now where I'm being forced to use a VPS with asterisk while I'd rather use a local install for testing/development. So I'm proxying my calls of my local asterisk server through the VPS |
| 15:21.20 | Greenlight | Why |
| 15:21.44 | *** join/#asterisk navaismo (~navaismo@189.191.236.128) |
| 15:21.46 | boom^time | because the inbound and outbound will only allow traffic from the VPS. I like to develop locally. |
| 15:22.03 | Greenlight | You need a new ITSP :) |
| 15:22.19 | boom^time | Agreed. |
| 15:22.42 | Greenlight | However, from a technical point of view, it should work (I've done similar things before to get around routing issues) |
| 15:23.10 | Greenlight | Start with directmedia=no |
| 15:23.13 | boom^time | Great! Maybe you could give me some pointers. Like I said so far I can make outbound np, it'l connect, but then just one way audio. |
| 15:23.33 | Greenlight | Ensure all your entries in sip.conf have directmedia=no |
| 15:23.45 | boom^time | okay I'll add it |
| 15:24.00 | boom^time | just on the proxy? |
| 15:24.08 | Greenlight | And your dev box |
| 15:24.28 | Greenlight | You want to ensure that the RTP is going ITSP <--> proxy <--> devbox |
| 15:24.40 | Greenlight | Rather than trying to go direct,and likely hitting NAT issues |
| 15:26.16 | boom^time | Didn't help unfortunately. |
| 15:27.00 | boom^time | I've limited the port range for my local dev box for rtp to 12000-13000 and opened the udp ports on my firewall to point at it. verified the rules took with netcat. Still not making it through though. |
| 15:27.13 | Greenlight | lets see your sip.conf |
| 15:27.31 | *** join/#asterisk CeBe (~CeBe@port-92-206-118-11.dynamic.qsc.de) |
| 15:30.06 | boom^time | http://pastebin.com/LPX86ncd |
| 15:31.54 | *** join/#asterisk Vico100 (dce97771@gateway/web/freenode/ip.220.233.119.113) |
| 15:31.58 | Vico100 | hello |
| 15:32.28 | Vico100 | i get this error: WARNING[1042]: db.c:285 db_execute_sql: Error executing SQL: database is locked |
| 15:32.36 | Vico100 | any help? |
| 15:32.51 | Vico100 | whenever device trys to register |
| 15:33.17 | wdoekes | Vico100: is your astdb writable? |
| 15:33.43 | Vico100 | im running asterisks on root and its set to |
| 15:33.56 | Vico100 | rwxr-xr-x |
| 15:34.33 | wdoekes | another asterisk process running? an open sqlite connection? |
| 15:34.56 | Vico100 | i think im using sqlite3 |
| 15:35.18 | Vico100 | i search google with this error not much can find |
| 15:35.23 | Vico100 | asterisk v11 |
| 15:35.25 | boom^time | Greenlight, http://pastebin.com/LPX86ncd (in case you didn't see it) |
| 15:35.43 | *** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir) |
| 15:37.58 | Greenlight | Your proxy doesn't have a [general] section ? |
| 15:38.30 | boom^time | no sorry it does, it's just the default sip.conf and the ITSP that set it up just added an include at the end pointing to what I pasted |
| 15:38.41 | boom^time | Do you want me to copy the whole thing? |
| 15:39.04 | Greenlight | Yes please |
| 15:40.07 | boom^time | Greenlight, here you go http://pastebin.com/Z8muC8y8 |
| 15:44.25 | Greenlight | I would try adding nat=force_rport,comedia to sip.conf |
| 15:44.47 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:44.47 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:45.01 | boom^time | to all of them? |
| 15:45.14 | Greenlight | No, just to the [general] section on the proxy |
| 15:45.21 | boom^time | okay |
| 15:45.28 | Greenlight | Might have no effect (am not sure what the default is) |
| 15:45.47 | Greenlight | Then restart and retry |
| 15:46.20 | boom^time | that did it |
| 15:46.31 | Greenlight | Oh, excellent :) |
| 15:46.33 | boom^time | now I need to know why lol |
| 15:46.49 | Greenlight | The default is presumany to not be nat aware |
| 15:47.01 | boom^time | I'm forwarding the ports directly to this machine though, so NAT shouldn't be an issue I wouldn't think |
| 15:47.02 | Greenlight | And your dev box is behind a nat ... |
| 15:47.06 | Greenlight | Well |
| 15:47.22 | Greenlight | Your dev box is sending an invite saying "hey I'm on 192.168.1.2" |
| 15:47.36 | Greenlight | And your proxy is taking that as gospell |
| 15:47.40 | boom^time | Gotcha |
| 15:48.00 | Greenlight | So the settings we added, tell it to instead use the IP it actually got the invite from |
| 15:48.26 | Greenlight | nat=force_rport,comedia generally work well in all situations |
| 15:48.48 | boom^time | that would have been a red flag for me except that I kept seeing this on the proxy "Probation passed - setting RTP source address to 75.151.20.174:12104" |
| 15:48.54 | boom^time | which is the WAN IP of the dev box |
| 15:49.09 | boom^time | if I would have seen 10.1.10.150:12104 then it would have dawned on me. |
| 15:49.15 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 15:50.08 | Greenlight | :) |
| 15:50.27 | boom^time | Well anyhow thanks a lot for the help. Now I get to go remove all of the settings I added to try and get this to work, whatever they were :) |
| 15:50.48 | Greenlight | No problem |
| 15:51.48 | *** join/#asterisk imox (~imox@91-64-148-46-dynip.superkabel.de) |
| 15:54.40 | *** join/#asterisk Draecos (~Draecos@124-149-73-161.dyn.iinet.net.au) |
| 15:58.29 | *** join/#asterisk eddic (~eddie@68.151.194.32) |
| 15:59.11 | navaismo | talking about force_rpot&comedia in the sip debug & sdp negotiation the contact address still the LAN address but sip show peers show the correct wan address |
| 16:00.05 | Greenlight | hm yea your missign some nat settings on dev box as well |
| 16:00.28 | Greenlight | You neec externip = <your wan address> |
| 16:00.30 | Greenlight | *need |
| 16:00.49 | Greenlight | and localnet=10.1.10.0/255.255.255.0 |
| 16:00.53 | eddic | Alright, so, I have an extension called "John" in the "internal" context. Inside a "pstn" context I go, "exten=>XXXXXXXXXX,1,Dial(SIP/John@internal)". Unforunately the Dial function tries to do name resolution on "internal" instead of know it is a context. What am I doing wrong? |
| 16:00.57 | Greenlight | (Or whatever your local network is ) |
| 16:01.09 | Greenlight | *and* nat=force_rport,comedia |
| 16:01.52 | Greenlight | eddic: That's not how you use Dial |
| 16:02.17 | Greenlight | Its wanting a dialstring |
| 16:02.23 | WIMPy | But maybe what you use Goto() for. |
| 16:02.29 | [TK]D-Fender | eddic: "john" is not in a "context" |
| 16:02.29 | eddic | Greenlight, So how do I "dial" an extension across a context like that? |
| 16:02.37 | *** join/#asterisk troyt (~troyt@2001:1938:240:2000::3) |
| 16:02.40 | WIMPy | Anyway, you're mixing up extensions and devices. |
| 16:02.51 | Greenlight | Either GoTo or you can use a Local channel |
| 16:03.04 | [TK]D-Fender | eddic: "context" is a term related to the dialplan |
| 16:03.15 | eddic | Hmmm. |
| 16:03.16 | eddic | Goto |
| 16:03.16 | WIMPy | Or maybe you can just include that other context. |
| 16:03.22 | eddic | Yeah, that makes more sense here |
| 16:03.26 | [TK]D-Fender | eddic: back up... |
| 16:03.48 | [TK]D-Fender | eddic: what is the name of the entry you made iin sip.conf? |
| 16:03.59 | navaismo | Greenlight, the sip.conf has the externip & localnets |
| 16:04.28 | Greenlight | Not the one he showed me |
| 16:04.48 | eddic | Goto worked like a charm |
| 16:04.50 | eddic | Thanks! |
| 16:06.31 | *** join/#asterisk tuxx- (tuxx@2a02:2308::216:3eff:feac:73b6) |
| 16:07.40 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 16:07.51 | *** join/#asterisk italorossi (~italoross@67.201.69.130) |
| 16:08.07 | tuxx- | hi. i'm writing a script for nagios that checks the asterisk P.I.D. to see if asterisk has quit. My question is, does asterisk spawn child processes which have a different P.I.D. from the main asterisk process? A couple of times per day my nagios script will say the P.I.D. changed, but when i check asterisk the uptime tells me that its still running and hasnt quit yet. |
| 16:08.28 | *** join/#asterisk threesome (~threesome@ip-94-113-12-233.net.upcbroadband.cz) |
| 16:10.00 | Greenlight | if you run an asterisk console you'll get another asterisk pid |
| 16:10.05 | Greenlight | COuld that be effecting things ? |
| 16:10.41 | tuxx- | my script should avoid those things |
| 16:10.54 | tuxx- | i iterate through all /proc/<num>/stat files and see which one has (asterisk) in it |
| 16:11.02 | tuxx- | so i dont think asterisk cli's will be picked up by my script |
| 16:11.09 | Greenlight | Why not > |
| 16:11.18 | WIMPy | +1 |
| 16:11.35 | drmessano | Seems like they absolutely would be |
| 16:11.42 | WIMPy | Why don't you do the good old killall -0 asterisk? |
| 16:12.27 | tuxx- | im gonna test that asterisk cli thing Greenlight / drmessano |
| 16:12.45 | WIMPy | Or that is the point where it helps if you use rasterisk instead of asterisk -r. |
| 16:12.58 | *** join/#asterisk italorossi (~italoross@67.201.69.130) |
| 16:13.02 | Greenlight | I guess the bigger question is why is your asterisk quiting ? |
| 16:13.11 | tuxx- | you guys are right |
| 16:13.24 | tuxx- | thats not my task Greenlight, sadly. |
| 16:13.35 | WIMPy | And the reason I use rasterisk -R so i kann killall -9 asterisk if I need to without killing the console. |
| 16:13.57 | tuxx- | but the console will be killed when you kill asterisk right? |
| 16:13.59 | tuxx- | ; |
| 16:13.59 | tuxx- | :P |
| 16:14.11 | WIMPy | Not hat way, no. |
| 16:14.37 | WIMPy | -R vs -r |
| 16:15.28 | WIMPy | So the r in front seems senseless, but keeps you safe from the killall. |
| 16:16.11 | Greenlight | And it reconnects when asterisk is back up? |
| 16:16.33 | WIMPy | As soon as it starts. |
| 16:16.46 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 16:16.50 | WIMPy | Unless you take more thna 30s to do so. |
| 16:17.05 | Greenlight | Nice, I'm going to start using that |
| 16:17.14 | WIMPy | So you see the whole startup process in the remote console. |
| 16:17.28 | Greenlight | Yup that's what I was thinking, very handy |
| 16:17.40 | WIMPy | Well, maybe you use the first few lines. But it seems to start really quick. |
| 16:18.06 | WIMPy | s/use/lose/ |
| 16:24.54 | *** join/#asterisk italorossi (~italoross@67.201.69.130) |
| 16:28.16 | *** part/#asterisk eddic (~eddie@68.151.194.32) |
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| 16:56.41 | *** join/#asterisk floren (~floren@unaffiliated/floren) |
| 16:58.53 | eddic | So I'm having trouble understanding something conceptually. I've got an asterisk server with a public IP on one NIC, and the natted lan on the the other nic. I've got a snom 320 on the lan. If I dial something like sip:user@someotherdomain.com on my phone, does that have anything to do with the asterisk server? Does it completely bypass the server and handle things on it's own? |
| 17:01.29 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:01.29 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:01.52 | navaismo | have you tried that? what show your cli? |
| 17:03.37 | *** join/#asterisk geek (~killown@unaffiliated/geek) |
| 17:08.40 | navaismo | If your phone is registered only with your asterisk and you send a sip uri you need to handle that via your dialplan too(as far I know). |
| 17:11.22 | eddic | Hmmm |
| 17:11.45 | eddic | Is there some sort of test sip uri people can call that maybe echo's back or something? |
| 17:15.26 | navaismo | not sure if this work but is using in blink-->echo@conference.sip2sip.info |
| 17:16.15 | [TK]D-Fender | eddic: on YOUR server? |
| 17:16.39 | [TK]D-Fender | eddic: only if YOU create it in your dialplan |
| 17:17.21 | [TK]D-Fender | eddic: also in general start throwing away the concept of "domains" |
| 17:19.18 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.45) |
| 17:20.32 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.45) |
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| 17:24.21 | *** part/#asterisk eddic (~eddie@68.151.194.32) |
| 17:29.15 | blehxor | Whats your favorite open source softphone? |
| 17:29.53 | blehxor | (linux) |
| 17:30.35 | navaismo | linphone or sflphone |
| 17:30.46 | navaismo | but you can use jitsi also |
| 17:31.55 | blehxor | thanks |
| 17:35.49 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:35.49 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 18:05.59 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 18:29.39 | *** join/#asterisk Rahail (~Rahail@67.214.121.181) |
| 18:29.45 | Rahail | Hi there I am using asterisk realtiem with mysql |
| 18:30.00 | Rahail | my mysql load like 500% and it cuase call ahng and lot of other problem |
| 18:30.10 | Rahail | is there any optomizition for asterisk with mysql i can update |
| 18:35.34 | *** join/#asterisk kenlik (~alberto@187.23.84.230) |
| 18:40.37 | kenlik | Hi, i'would like some help with the problem described in => http://pastebin.com/nmXxvnts |
| 18:41.58 | igcewieling | kenlik: have you read the Asterisk Book/ |
| 18:42.00 | igcewieling | ? |
| 18:42.02 | WIMPy | Amy ways to do that. |
| 18:42.29 | WIMPy | Either just a static 2nd try or the big one with registercontexts and dundi. |
| 18:42.34 | Rahail | igcewieling any idea about how can stop that mysql load |
| 18:42.35 | Rahail | that high |
| 18:42.41 | navaismo | ins't better to use a sip proxy to do that? |
| 18:42.48 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 18:42.54 | igcewieling | Rahail: no, but sounds like a Linux issue, not an Asterisk issue |
| 18:42.59 | kenlik | igcewieling, not; sorry but i'm newbie with asterisk |
| 18:43.08 | igcewieling | kenlik: then you should go read the Asterisk Book. |
| 18:43.10 | igcewieling | ~book |
| 18:43.10 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:46.01 | WIMPy | kenlik: And you need to learn the difference between extensions and devices. |
| 18:47.55 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:47.55 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:48.42 | kenlik | WIMPy, igcewieling: extensions and devices? Good start point, thanks |
| 18:57.03 | *** join/#asterisk ChadArag_ (~ChadArago@63-235-131-194.dia.static.qwest.net) |
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| 19:06.26 | *** join/#asterisk zewelor (~x@unaffiliated/zewelor) |
| 19:08.47 | zewelor | Hi, my asterisk instalaltion on ubuntu server was working fine few months, and now when i call someone / or someone calls me, i cant hear anything on both ends. I can't see anything obvious in logs using asterisk -vr. How can I debug it ? |
| 19:11.01 | *** join/#asterisk Praise- (~Fat@unaffiliated/praise) |
| 19:11.59 | igcewieling | zewelor: NAT or direct media or codec issue. |
| 19:12.11 | igcewieling | see sip.conf.sample for NAT info |
| 19:13.19 | zewelor | but without working NAT I think there will be no rining at all ? |
| 19:13.52 | igcewieling | maybe, maybe not. ringing doesn't have to use audio |
| 19:20.44 | zewelor | strange is I didnt change asterisk configuration since few months |
| 19:21.18 | zewelor | and I'm using same grandsteam voip gateways, which were working fine for last 2 years |
| 19:21.31 | *** join/#asterisk jalewis (~jlewis@solo.atlantic.net) |
| 19:25.20 | zamba | igcewieling: i guess i'm my own provider here.. (regarding the follow-up question about SPA-112 and fax) |
| 19:25.39 | jalewis | anyone aware of issues with dahdi 2.7.0 and latest centos 6.4 x86_64? Trying to setup a new server, and it crashes as soon as the wcte13xp driver loads: http://pastebin.com/Je9NgVhL |
| 19:25.46 | zamba | igcewieling: i'm not running SIP towards my provider.. that's a BRI (PRI?) interface.. isdn |
| 19:30.03 | WIMPy | zamba: You try to send a fax from an analog port to an ISDN line via VOIP? |
| 19:33.13 | WIMPy | wonders if that was a topic from several weeks ago. |
| 19:33.37 | *** join/#asterisk serafie1 (~erin@nat/digium/x-pztdwqbxnqvtsvpz) |
| 19:33.41 | igcewieling | WIMPy: days, not weeks |
| 19:39.53 | zamba | WIMPy: true that :) |
| 19:40.25 | zamba | today we have an alcatel pbx.. which terminates 4 isdn channels |
| 19:40.36 | zamba | is that 2 PRI or 2 BRI, i never remember :) |
| 19:40.39 | zamba | PRI, i believe? |
| 19:41.05 | WIMPy | BRI= 2channels, PRI = 30 channels |
| 19:41.06 | zamba | and this pbx has some analogue lines.. and the fax is connected to this.. we want to replace this alcatel pbx with asterisk |
| 19:41.13 | WIMPy | Or 24/23 on T1. |
| 19:41.13 | zamba | ah, ok.. 2 x BRI, then |
| 19:41.54 | WIMPy | You don't want to connect a fax machine via an VOIP ATA. |
| 19:42.54 | zamba | suggestions? |
| 19:43.04 | zamba | because the alcatel has to go |
| 19:43.27 | WIMPy | Use an ISDN to analog converter. |
| 19:43.41 | WIMPy | Oder at least an analog card. |
| 19:43.53 | zamba | which then will just msn listen for that one fax number? |
| 19:43.58 | zamba | or whatever it's called |
| 19:44.32 | WIMPy | If you have MSNs that is. |
| 19:44.49 | WIMPy | If you have DDI you will have to take it through the server first. |
| 19:45.33 | zamba | not sure what i do have |
| 19:51.42 | WIMPy | A liking for trouble, I guess. |
| 19:52.05 | WIMPy | Why do you want to replace the whole thing? |
| 19:54.00 | *** join/#asterisk bsaggy (~bsaggy@unaffiliated/bsaggy) |
| 19:56.17 | *** part/#asterisk zewelor (~x@unaffiliated/zewelor) |
| 19:59.17 | zamba | because i want the flexibility of using asterisk instead |
| 19:59.43 | zamba | more intelligent extensions |
| 20:00.09 | zamba | or do you suggest i do some programming of the alcatel pbx instead? and just use that as an isdn to analog converter? |
| 20:00.29 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 20:00.49 | WIMPy | That would be rather big converter, but certainly possible. |
| 20:01.42 | zamba | WIMPy: but what are your thoughts instead? |
| 20:02.22 | WIMPy | I don't know what you want to do you can't do now, so I can only comment on that fax thing so far. |
| 20:04.56 | anonymouz666 | anyone in here already used 3 digium cards in the SAME server? |
| 20:05.19 | igcewieling | anonymouz666: yes, but not in years |
| 20:05.34 | anonymouz666 | what models? |
| 20:06.47 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.40) |
| 20:10.12 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 20:10.18 | igcewieling | old analog cards |
| 20:10.40 | igcewieling | I assume a TDM400P since that is the only analog card Digium sold in 2002 |
| 20:17.22 | Katty | weeeeeeeeeeeeee |
| 20:17.47 | Katty | is putting asterisk 11.5 on vmware 5.1 this afternoon |
| 20:18.01 | Katty | and as usual, debian keeps mucking up my package names. |
| 20:18.06 | Katty | shakes fist |
| 20:19.20 | leifmadsen | maybe it's vmware |
| 20:20.10 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
| 20:20.13 | Katty | nah. |
| 20:20.31 | Katty | they just keep renaming the packages and i have to go digging through the cache trying to find the new name |
| 20:20.40 | Katty | you leave my vmware out of this mister! |
| 20:21.38 | leifmadsen | does not approve of vmware usage :) |
| 20:21.43 | leifmadsen | openstack ftmfw! :) |
| 20:21.50 | Katty | pats leifmadsen |
| 20:21.57 | Katty | i do many, many things you would no doubt disapprove of |
| 20:22.05 | leifmadsen | oh I like where this is going |
| 20:22.19 | leifmadsen | go on |
| 20:22.42 | Katty | frowns |
| 20:22.50 | Katty | shouldn't you be trying to weasel schnitzle out of file? |
| 20:23.00 | file | O.O |
| 20:23.15 | Katty | checks file's office for leftover schnitzle |
| 20:23.30 | Katty | file: OHOH, i made poutine last night. with the beef and gravy thing you suggested. |
| 20:23.38 | file | Katty, anddddddd? |
| 20:23.41 | Katty | file: it was good ^__________^ |
| 20:23.48 | file | Katty, :D |
| 20:24.17 | Katty | waits around on make |
| 20:24.49 | Katty | this intentionally takes forever so i have time to pluck my eyebrows. |
| 20:24.51 | Katty | RIGHT?! |
| 20:26.16 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 20:26.38 | Katty | aww lookit |
| 20:26.44 | Katty | a shiny new asterisk box ^_^ izzopretty |
| 20:26.51 | Katty | NOW I GET TO MUCK IT UP |
| 20:30.20 | [TK]D-Fender | #canthavenicethings |
| 20:31.19 | *** join/#asterisk CeBe (~CeBe@port-92-206-118-11.dynamic.qsc.de) |
| 20:33.46 | Katty | well. now what. |
| 20:34.03 | Katty | i should totally get some channels. |
| 20:35.16 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:dd08:6ef8:bec2:d0dd) |
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| 20:50.35 | igcewieling | Has anyone ever heard of a carrier doing calling waiting on SIP using INFO packets? Sounds crazy, but here is a packet http://pastebin.ca/2429668 |
| 20:51.29 | *** join/#asterisk neonerz (~g_bove@cpe-024-163-104-016.nc.res.rr.com) |
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| 20:54.02 | *** join/#asterisk fprior (c9dc96b2@gateway/web/freenode/ip.201.220.150.178) |
| 20:55.42 | WIMPy | Looks interesting. |
| 20:58.36 | fprior | Hi all. I read in sip.conf: "Note that direct T.38 is not supported." in case of directmedia=yes. This mean I cannot send/receive T.38 faxes if I set directmedia=yes ? |
| 20:59.36 | file | it means that T.38 will always goes through Asterisk, it will never go directly |
| 21:00.30 | igcewieling | file: any reason for that? |
| 21:00.38 | fprior | @file. In my situation, since I've set directmedia=yes, I cannot receive T.38 faxes |
| 21:00.59 | file | igcewieling, the UDPTL stack in Asterisk tends to make implementations less stupid |
| 21:01.13 | igcewieling | fprior: we have never been able to do t.38 when direct media is enabled. |
| 21:01.41 | igcewieling | which kind of sucks since we use the same peers for voice and fax |
| 21:02.04 | fprior | igcewieling this is not a good news |
| 21:08.06 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
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| 21:50.14 | eddic | If one has an audio file. How exactly does one make this file usable for Background() playback on an asterisk system? |
| 21:51.54 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 21:52.00 | *** part/#asterisk eddic (~eddie@S0106002590c4f7da.ed.shawcable.net) |
| 21:52.57 | [TK]D-Fender | have it in a format * has a format_XXX module for |
| 21:53.12 | [TK]D-Fender | they are all well documented |
| 22:28.55 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 22:30.25 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
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| 22:49.14 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 22:53.55 | WIMPy | Great. A sip to sip call where Asterisk is transcoding. But with one-way-audio. |
| 22:54.02 | WIMPy | Is that a new feature? |
| 22:56.47 | [TK]D-Fender | doesn't mean packets are making it both directions' |
| 22:57.11 | [TK]D-Fender | it simply negotiated WHAT you are losing |
| 22:57.12 | WIMPy | The phone said it received data. |
| 22:57.41 | WIMPy | And after transferring the call to a real phone I did get audio. |
| 22:59.27 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 22:59.56 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
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| 23:23.59 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 23:24.56 | Katty | pica |
| 23:25.06 | paulc | chew! |
| 23:25.15 | Katty | paulc++ |
| 23:25.37 | paulc | (or Pika.. those guys who made Dialogic-competing voice boards, back in the day? Good Canadian company..) |
| 23:26.57 | ChannelZ | gesundheit |
| 23:27.41 | WIMPy | And who also had theyr own asterisk channe. |
| 23:27.46 | WIMPy | l |
| 23:27.55 | [TK]D-Fender | Sangoma is a nice Canadian company.... |
| 23:28.19 | ChannelZ | Justin Bieber is from Canada |
| 23:28.30 | [TK]D-Fender | Nobody's perfect... |
| 23:28.33 | ChannelZ | The karma on that offsets a whole lot of good. |
| 23:28.35 | carrar | KATTYROOO |
| 23:29.04 | [TK]D-Fender | ChannelZ: Honey Boo-Boo < don't make this a war.... |
| 23:29.29 | ChannelZ | I wish we could export her to Canada as payback |
| 23:30.05 | Katty | hugs on carrar |
| 23:30.42 | carrar | w00t! |
| 23:31.15 | Katty | how's miss peggy |
| 23:31.47 | carrar | Katty: http://www.youtube.com/watch?v=cg2M-DHA-Js |
| 23:31.58 | carrar | in case you need a wtf work safe video |
| 23:33.33 | Katty | watches |
| 23:34.17 | Katty | confused, keeps watching |
| 23:34.23 | carrar | heh |
| 23:35.40 | Katty | oh my. |
| 23:35.44 | Katty | just closes tab |
| 23:35.49 | carrar | heh |
| 23:35.57 | carrar | yummie huh? |
| 23:35.57 | Katty | frownyfaces at carrar |
| 23:35.59 | carrar | WHAT |
| 23:36.02 | Katty | teehee |
| 23:36.06 | carrar | It's the hip new way to drink! |
| 23:36.21 | Katty | apparently! |
| 23:36.23 | carrar | heh |
| 23:36.44 | carrar | I mean what part of "drinking out of a toilet" doesn't sound appealing |
| 23:37.04 | carrar | with a straw no less |
| 23:37.10 | carrar | kids love it! |
| 23:37.48 | Katty | i'd say just about all of it doesn't sound appealing :P |
| 23:37.55 | carrar | heh |
| 23:47.41 | ChannelZ | This is better http://youtu.be/tLt5rBfNucc |
| 23:54.56 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 23:59.38 | WIMPy | So does anyone have an idea how this one-way-audio while transcoding thing might work? It seems to have come with the new bridging stuff. |