06:59.57 | *** join/#asterisk infobot (~infobot@rikers.org) |
06:59.57 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0-beta1 (2010/07/23), 1.6.2.10 (2010/07/23), 1.4.34 (2010/07/23), *-Addons 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.1 (2010/05/25), dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
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08:26.35 | Mark22 | Hello, I have 2 asterisk servers (one in a datacenter with a public IP (no NAT) and one in an office behind a NAT on a DSL connection). The one in the office registers with the one in the datacenter. However we have the problem that one of them says that the other down is and the connection isn't restored automatically. How can I find what the problem is and what the solution is? |
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08:31.08 | zoa | could you paste how it says that ? |
08:32.17 | zoa | does you office connection have a fixed ip ? |
08:34.51 | Mark22 | http://yourpaste.net/5855/ << that is on the datacenter side, at the office asterisk server nothing is shown |
08:35.02 | Mark22 | the connection does have a fixed IP |
08:36.04 | Mark22 | after shutting down the asterisk server at the office and waiting for 5 minutes and starting that asterisk server again it works (but for how long is the question and I want to solve it for the future if possible) |
08:37.35 | *** join/#asterisk suneeel (~suneeel@115.252.84.170) |
08:45.07 | zoa | ok so the connection is actually gone |
08:45.20 | zoa | do you have an idea what type of nat it is ? |
08:46.21 | zoa | did you put a qualify in sip.conf ? |
08:48.36 | zoa | look up the nat timeout for your firewall and make sure the value in qualify=$value is less |
08:48.49 | zoa | that should keep the connection up |
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08:49.59 | Mark22 | it is nat on a simple dlink modem, the connection is gone (but only 1 asterisk server "knows" it is gone and the other says the connection is up). at both servers I have set (in sip.conf): qualify=yes |
08:51.06 | zoa | decrease this value "defaultexpiry" on both servers to be under 30 seconds |
08:51.15 | zoa | i think your nat is timing out |
08:52.24 | zoa | qualify=30000 should work as well with less load on the server (as long as the ip doesnt change on the office side) |
08:52.53 | zoa | oh wait |
08:52.57 | zoa | i might be saying stupid things there |
08:53.48 | zoa | qualifyfreq is the value you want to change |
08:53.52 | zoa | not the qualify value |
08:54.17 | suneeel | wondering if anyone here has faced call progress detection problems with the tdm400p board |
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08:55.10 | zoa | Mark22, without changes the sip connection should go back up after 120s though i think |
08:55.16 | suneeel | My dialplan starts executing way before the other end picks up an outbound call... |
08:55.53 | Mark22 | it did re-register, however it didn't work :S |
08:55.56 | suneeel | Am I asking this in the correct place? |
08:56.01 | zoa | suneeel, how does you dialplan look like ? |
08:56.44 | zoa | Mark22, asterisk will retry the qualify every 60 seconds |
08:56.51 | Mark22 | zoa: should I place qualifyfreq=30 in sip.conf under the general section? |
08:56.52 | zoa | and the reregistration every 120s i think |
08:57.05 | zoa | should work there |
08:57.11 | zoa | but doesnt need to be there i think |
08:57.24 | suneeel | well, I'm actually issuing an originate over AMI and directing it to an AGI script that plays a prompt to the callee. But the AGI starts executing before the called party picks up |
08:57.29 | zoa | i'd start with the defaultexpiry and turn of the qualify |
08:57.33 | Mark22 | i did see that, it did register for 2 minutes and after that was unreachable for 1 minute (but in reality it didn't work at all) |
08:57.47 | zoa | i think you now have |
08:57.53 | zoa | nat timeout after 30 seconds |
08:57.57 | zoa | reregister after 120 |
08:57.58 | suneeel | I read on one of the discussion boards that this is because the 400p assumes that the call is connected as soon as the exchange picks up.. |
08:58.01 | zoa | and qualify after 60 |
08:58.19 | zoa | which means that even if the register worked, it will take up to 60 seconds before the qualify will approve it |
08:58.29 | zoa | and by then the nat might have timed out again |
08:58.37 | suneeel | So I'm guessing i need to set up the correct call processing tones in the config in order to detect the remote end off hook |
08:58.58 | zoa | suneeel, i wont be able to help you with that one im affraid |
08:59.42 | suneeel | damn! well thought it was worth a try in any case.. Any idea where I could look for more info? |
09:00.11 | Mark22 | could it be a solution to use port forwarding and stop using register (but just configure the servers at both ends)? |
09:00.22 | zoa | i'd say stick around and ask here again in an hour or so |
09:00.43 | zoa | Mark22, i wouldnt do that, you would need to do too much forwarding for the rtp |
09:00.51 | suneeel | btw.. The same behaviour is exhibited when I put use a dialplan context instead of an AGI.. Which is why i said dial plan i the first place |
09:01.04 | suneeel | Oh ok |
09:01.06 | suneeel | Cool |
09:01.08 | zoa | aha |
09:01.08 | suneeel | Will do that |
09:01.23 | zoa | so its not ami related (I'm not familiar with AGI) |
09:01.52 | zoa | suneel can you paste the dialplan part somewhere and the cli output ? |
09:02.22 | suneeel | k, gimme a sec |
09:03.08 | zoa | use pastebin please |
09:03.15 | hrhrhr_ | anything worth shouting about in 1.8? |
09:04.08 | suneeel | Ok, someone's gone and rebuilt my asterisk box.. Should have checked that before comming on here I guess |
09:04.25 | suneeel | Will get back to you guys tommorow |
09:04.28 | suneeel | Thanks |
09:04.49 | Mark22 | and it isn't working anymore :( I need to fix it today or we will loose this client :S |
09:06.44 | zoa | Mark22, just try the defaultexpiry, i think it will work |
09:07.01 | Mark22 | I did already change that |
09:07.06 | zoa | no luck ? |
09:07.11 | Mark22 | no luck :( |
09:07.21 | zoa | that doesnt sound good |
09:07.29 | zoa | what dlink is it |
09:07.30 | zoa | ? |
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09:07.42 | zoa | check if is has some option for statefull packet inspection |
09:07.46 | zoa | or packet rewriting |
09:08.02 | zoa | some of the dlinks are known to fuck up sip |
09:08.17 | zoa | take a tcpdump on the server when you try dialing and see what it returns |
09:08.30 | Mark22 | I did just recheck it, it is a THOMSON ST546 if I may believe the web interface |
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09:08.52 | zoa | aha, thomson is better |
09:09.22 | zoa | if you get lucky you will see am icmp unreachable when trying to connect to the office pbx |
09:10.10 | zoa | if you want to rule out sip rewriting issues on the thomson (not sure if it does that, i dont know that model), use iax2 |
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09:14.15 | Mark22 | ICMP echo reply, id 53481, seq 27076, length 44 << looks normal to me, now I did get a "Auto fallthrough, channel 'SIP/1010-0000005e' status is 'CONGESTION'" in the logs |
09:14.27 | Mark22 | is iax2 more stable compared to sip? |
09:15.52 | zoa | not really, helps a bit for situations where you have no audio because of nat, but you get stuck before that |
09:17.00 | zoa | i'd remove the qualify completely for now, and just use the defaultexpiry |
09:17.03 | zoa | on both servers |
09:17.30 | zoa | and try iax2 if that doesnt work as you are in a hurry |
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09:43.01 | mpe | Hi do any of you know to set the timeout that is used with asterisk attende transfer *2 |
09:43.01 | mpe | at the moment the call timeout after 15 sec, where I get this message |
09:43.02 | mpe | features.c:1957 ast_feature_request_and_dial: We exceeded our AT-timeout |
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10:11.59 | garymc | is making my phone show 01412341234 instead of 1412341234 a really hard task to do? I can live without the 0 but it would be nice if I could get the phone to show the 0 |
10:12.17 | garymc | hence the full number |
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10:38.18 | zoa | where do you want to show that ? |
10:38.26 | zoa | you mean the caller id for incoming calls ? |
10:39.27 | zoa | http://www.voip-info.org/wiki/view/Setting+Callerid |
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11:03.28 | zoa | hierse puzzled |
11:03.32 | zoa | lang geleden |
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11:04.59 | puzzled | hey zoa. how are things? |
11:05.12 | zoa | goooood |
11:05.14 | zoa | and you ? |
11:05.57 | puzzled | besides a massive headache I'm good |
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11:19.23 | sputnick | hi there |
11:19.25 | puzzled | zoa: is zoiper communicator biz 64bits available for Fedora 13? I only see 32bits F10 releases on the website and F10 is EOL for quite some time |
11:20.23 | zoa | i will check just a sec |
11:21.38 | zoa | nopez, only for ubuntu apparently |
11:21.52 | zoa | will check to make a new one |
11:22.01 | zoa | we are a bit understaffed because of the holidays |
11:22.04 | zoa | i will put it on the todo |
11:22.09 | puzzled | no problem. thanks |
11:22.47 | sputnick | is there a particular setting to call a mobile number directly to a voicemail ? Because I have had a bug on a asterisk/nagios/dedicated server calling my mobile phone every 5 minutes but never ringing, my French provider Free ask me 89⬠this month just for this. |
11:24.11 | sputnick | $116 |
11:24.44 | puzzled | zoa: the Windows version of Communicator Biz is 32bit too? |
11:24.48 | zoa | yes |
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11:24.51 | sputnick | use SIP |
11:24.57 | zoa | doesnt really make a difference to run them in 64 bit |
11:25.06 | zoa | only for DSP Stuff |
11:25.25 | zoa | they should run fine in 32bit on 64 bit machines |
11:25.41 | puzzled | zoa: ok, ta |
11:25.59 | zoa | its a little bit faster, but not a lot and we usually dont use over 4gb of ram :) |
11:26.23 | zoa | sputnick |
11:26.30 | zoa | sip cannot do that |
11:26.38 | zoa | but, depending on your cellphone provider |
11:26.42 | zoa | there might be an option to do that |
11:26.55 | zoa | with some dtmf voodoo |
11:26.59 | zoa | or special numbers to dial |
11:27.01 | zoa | i have seen it before |
11:27.42 | zoa | you might have to call your cellphone from another cellphone from the same provider though to make it work |
11:27.58 | sputnick | zoa: I have no option from Free provider, nor special number, but I have a call every 5 minutes since june 26 that cost me a bit each |
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11:29.15 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
11:30.33 | zoa | you will need to check with your mobile operator |
11:30.49 | zoa | i doubt you can do anything on Asterisk to do that |
11:31.02 | zoa | ah look |
11:31.05 | zoa | its mister leif |
11:31.15 | puzzled | indeed |
11:31.52 | zoa | good morning sunshine! :P |
11:35.13 | leifmadsen | zoa: :) |
11:35.17 | leifmadsen | is now writing documentation |
11:39.31 | prgmrchris | sputnick: there has to be a way, http://slydial.com/ does exactly what you are describing and im sure they use asterisk to do it |
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11:44.51 | zoa | sputnik is from france |
11:47.05 | zoa | this is how to do it for one us provider: http://www.ehow.com/how_4787651_send-message-directly-voicemail.html |
11:47.19 | zoa | i imagine that slydial does such a trick for every provider |
11:49.05 | zoa | but why would you send it to voicemail ? You will still pay for it ? |
11:51.54 | *** part/#asterisk rushowr (~rushowr@99-28-31-100.lightspeed.stlsmo.sbcglobal.net) |
11:53.29 | prgmrchris | zoa: only thing is how does slydial know which provider a mobile number is on? with lnp i dont think thats easy unless you have some kind of inside info |
11:53.37 | prgmrchris | i doubt they are just bruteforcing it |
11:54.38 | zoa | number portability db or so :) |
11:54.48 | zoa | i investigated such a functionality for belgium before |
11:54.56 | zoa | and did not find a common api |
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12:01.52 | jkroon | hi guys, I've got three BRI (B410P) cards in ptmp mode, now they were up just now, except for two lines, then swapped the two dead ports with two others now all four is dead. |
12:01.54 | jkroon | any ideas? |
12:02.00 | jkroon | (dead => RED alarm) |
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12:23.16 | jkroon | ok, the only way to get them pack up is to reboot. |
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12:39.01 | DND | guys im getting segfault from asterisk: segfault at 000000000ee04000 rip 00002aaaad526d66 rsp 0000000040bc35a0 error 4 |
12:39.12 | DND | im not sure where it came from. i just typed "dmesg" |
12:39.22 | jkroon | can be anywhere. |
12:43.05 | [TK]D-Fender | DND: Going to tell us anything about your install? |
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12:54.51 | garymc | [TK]D-Fender : have you any idea how long it would take for a polycom software guy to decide if they would add RFC3326 or dismiss it? I know you dont work for polycom but would you have an educated guess? |
12:54.54 | *** join/#asterisk cusco (~trilili@33.83.136.95.rev.vodafone.pt) |
12:54.57 | cusco | hi all |
12:55.22 | garymc | as the feature request is open and active. just no notes on it |
12:55.42 | cusco | asterisk 1.6.2 on a debian stable machine is keeping a big list of opened file descriptors |
12:56.36 | [TK]D-Fender | garymc: No, and guessing is pointless |
12:57.02 | [TK]D-Fender | garymc: link me to the request page |
12:57.26 | [TK]D-Fender | cusco: And that isn't a version # |
12:58.56 | garymc | its passworded |
12:59.15 | garymc | https://jira.polycom.com:8443/browse/EXT-3029 |
13:00.08 | garymc | [TK]D-Fender : Have you got an account ? |
13:02.18 | [TK]D-Fender | garymc: No |
13:02.53 | garymc | [TK]D-Fender : does the link ask for a password? |
13:03.10 | [TK]D-Fender | garymc: indeed |
13:03.34 | garymc | [TK]D-Fender : Thats the end of that then |
13:05.34 | garymc | Just wondering if i would have to add much or how difficult would it be for me to get the 0 to show on my phones. When someone calls me from say 01412342345 my phone show 1412342345 |
13:06.00 | garymc | i know its no biggie, but just wondered how I would do it and what file I need to be in |
13:06.12 | leifmadsen | garymc: modify the CALLERID() ? |
13:06.15 | cusco | http://paste.debian.net/81729/ |
13:06.34 | cusco | [TK]D-Fender: Connected to Asterisk 1.6.2.9 |
13:06.49 | cusco | this was hapenning with 1.6.2.8 |
13:06.58 | leifmadsen | lots of file descriptors are normal especially if the system is under load |
13:07.01 | garymc | leifmadsen i dont know how to do it |
13:07.06 | [TK]D-Fender | garymc: It shows no 0 because there is no 0 coming in |
13:07.09 | leifmadsen | garymc: core show function CALLERID |
13:07.12 | [TK]D-Fender | garymc: Add one yourself |
13:07.29 | garymc | ok how do i add it? |
13:07.31 | [TK]D-Fender | leifmadsen: FreePBX <- |
13:07.39 | leifmadsen | [TK]D-Fender: oh then I'm done here |
13:07.45 | garymc | yeah but its an asterisk thing is it now |
13:07.48 | garymc | *not |
13:07.49 | [TK]D-Fender | garymc: No |
13:08.07 | leifmadsen | In Asterisk, I'd just do: Set(CALLERID(number)=0${CALLERID(number)}) |
13:08.08 | garymc | oh n3glv said it was an asterisk thing |
13:08.14 | cusco | leifmadsen: lots of udp connections, it seems that i doesn't terminate them after the calls |
13:08.14 | [TK]D-Fender | garymc: it is dialplan, not device config. |
13:08.26 | leifmadsen | cusco: check the bug tracker to see if that is an open issue |
13:08.28 | [TK]D-Fender | garymc: n3glv said a LOT of things. |
13:08.35 | garymc | its dial plan so it displays the incoming number with the 0? |
13:08.47 | garymc | you wernt there I dont think |
13:08.50 | [TK]D-Fender | garymc: You modify CallerID in the dialplan. |
13:09.01 | garymc | ok ill go over to freepbx |
13:09.10 | [TK]D-Fender | garymc: Probelm is you have to integrate this right to work with your GUI |
13:09.16 | [TK]D-Fender | garymc: And we don't want to deal with that./ |
13:09.28 | garymc | ok, so its alot of hassle for me, then for you too :P |
13:09.44 | [TK]D-Fender | garymc: Yes, you are a lot of hassle for me... |
13:09.49 | garymc | :P |
13:10.03 | garymc | you will grow to love me |
13:12.44 | *** join/#asterisk diegomad (~mad@190.147.221.78) |
13:12.51 | garymc | or maybe not |
13:12.54 | zoa | garymc, i asked you before and i sent you in the right direction |
13:12.57 | zoa | scroll up please |
13:13.05 | garymc | you did ? |
13:13.25 | zoa | <zoa> http://www.voip-info.org/wiki/view/Setting+Callerid |
13:13.26 | garymc | I got disconnected and had to reboot |
13:13.34 | zoa | aha |
13:13.42 | garymc | my chatzilla doesnt cache |
13:13.56 | garymc | not sure if i can get it too either |
13:14.22 | zoa | so its possible and you can either go read on the set command or you can cheat and listen to leif :) |
13:14.50 | garymc | that link says thats for outgoing only |
13:16.28 | *** join/#asterisk FlashDeluxe (~FlashDelu@static-87-79-94-28.netcologne.de) |
13:16.53 | FlashDeluxe | Hi, Ive got a problem, if i want to dial i get an error: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
13:17.06 | FlashDeluxe | any suggestions? |
13:18.20 | [TK]D-Fender | FlashDeluxe: Tell us useful details about what you've got, configs, and the actual failed call to look at |
13:20.50 | FlashDeluxe | Ok, thx :) i`ve asterisk 1.6.2.10 installed with dahdi-linux-2.3.0.1 and dahdi-tools-2.3.0 |
13:21.03 | FlashDeluxe | the configs are default |
13:21.21 | [TK]D-Fender | FlashDeluxe: PASTEBIN <- |
13:21.23 | [TK]D-Fender | ~pb |
13:21.24 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
13:21.32 | [TK]D-Fender | FlashDeluxe: "Default" is meaningless |
13:21.58 | garymc | leifmadsen : could I just add Set(CALLERID(number)=0${CALLERID(number)}) to sip_custom.conf file? |
13:22.01 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
13:22.09 | [TK]D-Fender | garymc: No |
13:22.11 | leifmadsen | garymc: probably not |
13:22.19 | [TK]D-Fender | DEFINITELY not |
13:22.22 | leifmadsen | garymc: I don't use FreePBX so I will be of no further assistance |
13:22.24 | garymc | ok |
13:22.26 | FlashDeluxe | okay. extensions.conf: [default] exten => 015771360963,1,Dial(DAHDI/g1/${EXTEN}) |
13:22.27 | [TK]D-Fender | garymc: This has nothing to do with SIP |
13:22.29 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
13:22.45 | garymc | ok |
13:22.48 | [TK]D-Fender | FlashDeluxe: PASTEBIN. You have multiple things to be showing us and you haven't told us what you're using./ |
13:22.54 | leifmadsen | FlashDeluxe: first of all, that's dangerous -- you should not allow outgoing calls via [default] |
13:22.54 | garymc | Is there I file I would add Set(CALLERID(number)=0${CALLERID(number)}) too? |
13:23.05 | leifmadsen | garymc: please use #freepbx -- we can't support it here |
13:23.15 | [TK]D-Fender | garymc: extensions(something).conf |
13:23.28 | [TK]D-Fender | garymc: And all sorts of other GUI changes to point things there |
13:23.32 | FlashDeluxe | its just for testing, i will change it later if it works |
13:23.37 | [TK]D-Fender | garymc: ... |
13:23.39 | [TK]D-Fender | ~wglwat |
13:23.39 | infobot | i guess wglwat is well, good luck with all that |
13:23.51 | [TK]D-Fender | garymc: 2nd door to your left. |
13:23.55 | FlashDeluxe | wait a minute i will show you my configs |
13:23.56 | prgmrchris | haha |
13:24.04 | garymc | im in there :S |
13:24.09 | prgmrchris | garymc is at it again |
13:24.24 | [TK]D-Fender | prgmrchris: Never really stops.. he just slows down a little on occasion |
13:24.53 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
13:25.21 | *** join/#asterisk Carp1 (~none@cpe-24-92-37-23.nycap.res.rr.com) |
13:25.41 | prgmrchris | [TK]D-Fender :) |
13:26.59 | drmessano | If I went to the pound and they had a dog that was missing its left legs, I would buy it, name it garymc, bring it home, set in the middle of the living room, and watch it run around in circles all day. |
13:27.03 | leifmadsen | #asterisk is not 2nd level #freepbx support though |
13:27.39 | garymc | ok |
13:28.01 | drmessano | leifmadsen: #trixbox Is Tier 2 Windows Vista support, however. |
13:28.17 | garymc | A dog missing its left legs would find it hard to stand up, why not try a boat with one paddles |
13:29.20 | drmessano | garymc: "try a boat with one paddles" <-- PLURAL, and a good indicator of your underlying lack of understanding |
13:29.30 | Carp1 | I have an old install of asterisk and when I choose 1 to update it says its timing out trying to reach the server. |
13:29.33 | garymc | typo |
13:29.39 | drmessano | PLURAL |
13:29.50 | [TK]D-Fender | Carp1: There is no such thing as "1 to update" |
13:30.05 | leifmadsen | sounds like a GUI problem |
13:30.07 | Carp1 | Whoops, I think it's AsteriskNOW |
13:30.16 | leifmadsen | 302 Redirect #asterisknow |
13:30.18 | [TK]D-Fender | Carp1: 3rd door to your left ... |
13:30.23 | Carp1 | Sorry. |
13:30.23 | drmessano | If I had a boat with one paddles, I would be up the creeks |
13:31.24 | [TK]D-Fender | drmessano: WITH a paddles! |
13:32.13 | drmessano | Sorry, I was typing with one hands |
13:32.32 | eppigy | good morning |
13:32.52 | eppigy | i too have challenges typing with one hand |
13:33.34 | drmessano | eppigy: Can you row a boat with one paddles? |
13:33.40 | drmessano | Then your in luck |
13:33.49 | garymc | drmessano was your other hand touching your vagina's ? |
13:34.10 | [TK]D-Fender | drmessano: Of course. I've canoed for almost 20 years :) |
13:34.18 | drmessano | garymc: No, she's at work |
13:34.27 | [TK]D-Fender | garymc: His vagina's what? |
13:34.38 | garymc | you have another half? whats his name? |
13:34.57 | drmessano | garymc: You're going nowhere with this. Stop while you're ahead |
13:35.15 | garymc | whats up, i thought this was "Lets take the piss time" |
13:35.45 | eppigy | i will never understand that term |
13:35.48 | drmessano | I'm not into golden showers |
13:35.49 | eppigy | takin the piss |
13:35.54 | drmessano | That's pretty gross |
13:35.55 | [TK]D-Fender | garymc: so far you're gender, plural, possessive, and grammatically challenged :) |
13:35.56 | garymc | its a uk thing |
13:36.01 | eppigy | oh i know |
13:36.11 | garymc | no need to gang up on me |
13:36.30 | garymc | [TK]D-Fender your not the other half thats at work are you? :P |
13:36.36 | eppigy | rude |
13:36.47 | [TK]D-Fender | garymc: "that's" |
13:36.48 | drmessano | [TK]D-Fender: You left off the part about being GUIetically challenged :) |
13:37.02 | drmessano | "You're" |
13:37.03 | garymc | ZZZZzzzzz |
13:37.24 | [TK]D-Fender | drmessano: Thought I'd leave you one ;) |
13:37.51 | eppigy | this mcdonalds coffee is actually pretty tasty |
13:37.56 | drmessano | I would call garymc on Skype and tell him to his face, but I don't have weeks to help him set it up |
13:38.03 | eppigy | loool |
13:38.11 | eppigy | burn |
13:38.13 | Carp1 | you got to goto "Stewarts Shops" for coffee |
13:38.20 | Carp1 | I think they're only in NY though |
13:38.20 | zoa | puzzled |
13:38.24 | zoa | was that for communicator ? |
13:38.31 | eppigy | Yeah I live in hotlanta |
13:38.33 | garymc | you got more than weeks , well you would need more than weeks |
13:39.01 | drmessano | garymc: Thank you for the props, mcgary. Word. |
13:40.01 | puzzled | zoa: yes the comm. biz pro/enterprise uber version |
13:40.43 | garymc | drmessano : word? like WORD UP! ? |
13:40.56 | garymc | OWW |
13:41.37 | [TK]D-Fender | garymc: Haven't you heard? |
13:41.43 | [TK]D-Fender | garymc: THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD... |
13:41.45 | [TK]D-Fender | ...BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE... |
13:41.46 | [TK]D-Fender | ...WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD!THE BIRD BIRD BIRD, THE BIRD IS THE WORD! |
13:41.53 | garymc | lol |
13:42.07 | garymc | I like Peter Griffins take on this wonderful song |
13:42.47 | garymc | [TK]D-Fender : ?pastebin |
13:42.54 | garymc | !pastebin |
13:43.15 | kaldemar | ~pb |
13:43.15 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
13:43.24 | kaldemar | [TK]D-Fender: ^^^^^ |
13:43.40 | [TK]D-Fender | kaldemar: Pardon? |
13:44.06 | [TK]D-Fender | kaldemar: that was 1 line which IRC's silly limit split in 3. Thus not OVER 3. FAIL :p |
13:44.39 | kaldemar | [TK]D-Fender: showed up as 11 on my tiny screen. but i was obviously just kidding. :) |
13:45.14 | *** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk) |
13:45.40 | kaldemar | i'll re-read on a 1800x1600 console. |
13:47.15 | puzzled | zoa: if it was not clear I mean the commercial version and very much prefer a 64bit version for F13 |
13:47.31 | zoa | ah we dont have biz versions for it i think |
13:47.42 | zoa | need to modify the shop a bit for that :/ |
13:47.47 | zoa | also on the todo :) |
13:47.55 | zoa | will see what i can do |
13:47.59 | puzzled | zoa: ok. i that case I'll settle for what you have that works on F13 x86_64 |
13:48.06 | puzzled | thanks |
13:48.08 | zoa | am already installing f13 here |
13:49.35 | *** join/#asterisk korcan (~johnynum5@ip65-44-169-66.z169-44-65.customer.algx.net) |
13:49.57 | coppice | does anyone here get a problem when using FAX for Asterisk that the FAX appears to complete OK, but the resulting TIFF contains only the first centimetre of the image? |
13:50.52 | joobie | http://tinyurl.com/36vp3fv |
13:51.21 | joobie | check it out |
13:51.29 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
13:51.37 | *** kick/#asterisk [joobie!~chatzilla@216.191.106.163] by [TK]D-Fender (joobie) |
13:51.43 | *** join/#asterisk joobie (~joobie@CPE-124-181-130-3.vic.bigpond.net.au) |
13:51.46 | joobie | :P |
13:51.56 | joobie | i take it u clicked it TK |
13:51.57 | [TK]D-Fender | joobie: Don't ... |
13:52.02 | joobie | heeh ok |
13:52.21 | joobie | you inspired me |
13:52.29 | joobie | with your THE BIRD BIRD BIRD THE BIRD IS THE WORD |
13:53.07 | FlashDeluxe | hi, can anybody show me his chan_dahdi.conf? :) |
13:53.26 | Carp1 | what ever happened to NuFone? I havn't been on the internet in a long time. |
13:53.29 | drmessano | Nice, NSFWOH |
13:54.18 | seanbright | welcome to 1998 |
13:54.28 | seanbright | he's been holding that pose a long time |
13:55.26 | drmessano | seanbright: Waiting.. |
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14:02.05 | FlashDeluxe | Hi@all: Ive got a problem: When i edit the chan_dahdi.conf by adding a 'channel => x-x' all of the dahdi commands are not longer available, they do not exist :( but as i delete the line 'channels => x-x' all of the dahdi commands do exist :S?? Any suggestions? My system is:asterisk 1.6.2.10 installed with dahdi-linux-2.3.0.1 and dahdi-tools-2.3.0 |
14:02.24 | seanbright | pastebin your entire chan_dahdi.conf file |
14:02.29 | seanbright | ~p |
14:02.30 | infobot | [p] q and not q |
14:02.30 | seanbright | ~pb |
14:02.31 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
14:03.58 | FlashDeluxe | here it is http://paste.debian.net/81737/ |
14:13.46 | chazzam | if you run 'dahdi_cfg -vvv' what gets turned up? |
14:14.25 | *** join/#asterisk patrick^ (~patrick_@dhcp-0-24-14-f5-ac-e2.cpe.mountaincable.net) |
14:16.17 | chazzam | FlashDeluxe: ^^ |
14:16.25 | FlashDeluxe | nothing |
14:16.27 | FlashDeluxe | DAHDI Tools Version - 2.3.0 |
14:16.36 | FlashDeluxe | thats all :( |
14:16.41 | chazzam | does it say "No channels to configure" or anything? |
14:16.56 | FlashDeluxe | no it says nothing^^ |
14:16.56 | chazzam | you have to configure the driver before asterisk can configure the channels |
14:17.29 | FlashDeluxe | damn...the maschine freezed |
14:17.41 | FlashDeluxe | i have configured it |
14:18.41 | chazzam | it sounds like you may have lower level problems than your asterisk config |
14:19.39 | FlashDeluxe | i guess sp -.- |
14:20.20 | [TK]D-Fender | [10:17]<FlashDeluxe>i have configured it <-- doesn't look like |
14:21.47 | chazzam | what card do you have? |
14:22.52 | FlashDeluxe | two no name hfcs [TK]D-Fender: But i configured it, but after that i installed a new asterisk version and since that it freezes |
14:23.18 | [TK]D-Fender | FlashDeluxe: show use "dahdi_cfg -vvvv" |
14:24.15 | FlashDeluxe | on moment plz, the machine is still booting |
14:24.22 | hrhrhr_ | when did they rename it to daddy |
14:25.31 | [TK]D-Fender | ~dahdi |
14:25.32 | infobot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel (more info at http://www.asterisk.org/dahdi ) and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
14:26.19 | FlashDeluxe | OK, after i executed the command i get this: http://paste.debian.net/81739/ |
14:27.03 | FlashDeluxe | and then the machine freezed |
14:27.11 | tzafrir_laptop | FlashDeluxe, segmentation fault by daudu_cfg? That's bad |
14:27.34 | chazzam | heh daudu |
14:27.36 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-17-NewEngland.hfc.comcastbusiness.net) |
14:27.49 | FlashDeluxe | i know^^ but what can i do? |
14:27.59 | *** join/#asterisk yabadabado (~root@static-213-115-44-229.sme.bredbandsbolaget.se) |
14:28.08 | yabadabado | hmm |
14:28.09 | tzafrir_laptop | There's a patch or two I got for bug reports |
14:29.52 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
14:31.15 | FlashDeluxe | tzafrir_laptop what for patches? |
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14:38.20 | Katty | guten morgan |
14:38.36 | *** join/#asterisk seanjohn (~seanjohn@gateways.sheltoncomputers.com) |
14:38.42 | FlashDeluxe | moin |
14:38.42 | seanjohn | chan_sip.c:13649 handle_response: Remote host can't match request CANCEL to call '176314020af796be26ef805e45dd962d@173.50.101.11 |
14:38.49 | seanjohn | morning |
14:38.58 | seanjohn | terrible yesterday and today |
14:39.15 | seanjohn | the audio quality went from perfect to choppy as hell |
14:39.23 | seanjohn | I see that in the damn manager |
14:40.05 | seanjohn | anyone know what's going on? |
14:40.31 | zoa | usually it means too much harddisk activity |
14:41.16 | seanjohn | ok zoa. that would help my diagnosis |
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14:41.34 | seanjohn | one of the hard drives in raid 5 is failing |
14:41.51 | seanjohn | I can't tell which one as noone can tell me any tool for linux to do a smart |
14:42.13 | seanjohn | otherwise, I would just remove it |
14:42.19 | zoa | is it hardware raid ? |
14:42.25 | seanjohn | nope |
14:42.28 | zoa | if so, ask the raid controller |
14:42.29 | zoa | ah damn |
14:42.50 | seanjohn | its only one bad sector |
14:42.56 | seanjohn | and linux is having a fit |
14:43.11 | zoa | hmm now that i think of it the raid 5 in software might already not be a very good idea |
14:43.23 | zoa | what are you doing that requires raid 5 ? do you monitor a lot ? |
14:43.35 | seanjohn | no, just for things like this |
14:43.57 | seanjohn | backups; its 4 drives in raid 5 and two can fail |
14:44.24 | zoa | k |
14:44.39 | seanjohn | but the damn thing won't go ahead and fail so that I get an error on post |
14:44.40 | zoa | which means you should be able to pull 1 more out and it should still work |
14:44.46 | *** join/#asterisk af_ (~getsmart@78.134.21.122) |
14:44.48 | zoa | so you could find the bad one with trial and error :) |
14:45.31 | seanjohn | i need to take the whole array out and use the backup drive but I think I need to ask others how to copy WHOLE partitions from one to another. I'm a windows geek |
14:45.59 | seanjohn | with the array out, I can test one at a time |
14:46.12 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-rtowfjbpcrsypxqt) |
14:47.04 | seanjohn | this isn't the proper channel for this |
14:47.11 | seanjohn | but I would appreciate some commands |
14:47.48 | seanjohn | tar up the whole drive? that's not going to restore the compiled programs and yum installs (centos 64) is it? |
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14:50.33 | kl4m | I have a question concerning ztmonitor |
14:50.45 | seanjohn | zaptel who uses zaptel anymore |
14:51.51 | anonymouz666 | those stuck on 1.2 |
14:51.56 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
14:53.07 | seanjohn | 1.4 is just a dialplan change away |
14:53.10 | seanjohn | lol |
14:54.41 | [TK]D-Fender | LOL...... NO |
14:55.19 | KavanS | sersly? |
14:56.03 | kl4m | so anyway, I have a local line and asterisk is sending milliwatt() over it. Shouldn't ztmonitor show ~14000 tx? it shows only 4600 |
14:56.03 | seanjohn | [TK]D-Fender: just joking |
14:56.25 | seanjohn | trying to keep myself from bashing my machine |
14:56.46 | seanjohn | NEW hard drives and one is failing??? western digital raptors |
14:57.10 | seanjohn | I thought the days of "1 in 10 fail within the first month" are gone |
14:58.12 | zoa | seanjohn i cant help you with that raid stuff |
14:58.15 | zoa | i only do hw raid |
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15:13.47 | Ken85 | hi. can you suggest any good voip cordless phones? I'm thinking of buying one |
15:13.48 | nny | pssh thank god |
15:14.19 | nny | tcpdump + wireshark + Telephony Tools + Graph Data = foolproof test to show that DTMF is being sent in RTP. Hooray! |
15:14.48 | WIMPy | Ken85: Don't go for wifi, go for dect. |
15:14.57 | nny | fun part is you can reconstruct calls and play them back too |
15:15.16 | nny | Snom M3 is ok, had some glaring support issues when one I had died though |
15:15.18 | Ken85 | WIMPy: yeah i dont prefer wifi although i dont know what dect is.. whats dect? |
15:16.21 | nny | http://en.wikipedia.org/wiki/Digital_Enhanced_Cordless_Telecommunications |
15:16.49 | Ken85 | nny Nice. do you have any recommendations? |
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15:17.28 | [TK]D-Fender | ~wifivoip |
15:17.29 | infobot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
15:17.33 | [TK]D-Fender | Ken85: ^^^ |
15:17.38 | nny | Ken85: I have used the snom m3 extensively. It's a bit light and simple interface. (Almost like a cheap cell phone) only issue I had was it took me 3 weeks to return one, their support process is a bit convoluted |
15:17.57 | nny | they have a new one out too I think, but haven't used it. |
15:18.02 | [TK]D-Fender | M3 = range and battery issues (from reports) |
15:18.11 | Ken85 | nny: how about the siemens gigaset? |
15:18.19 | nny | Ken85: haven't tried it |
15:18.20 | zoa | everythin on wifi is miserable on battery |
15:18.28 | nny | [TK]D-Fender: 150 feet is about right |
15:18.44 | nny | [TK]D-Fender: after my last support nightmare I am looking for a new one anyways |
15:18.47 | WIMPy | Ken85: If you're concerned aboput privacy, you could take a look at encryption used by various models on www.dedected.org. |
15:19.14 | nny | Ken85: if you get the siemens let me know, my business partner suggested it recently |
15:19.31 | Ken85 | nny: i want something for home use |
15:20.02 | Ken85 | just a phone which i ll be able to do my landline calls with normal phone and also be able to do sip phone calls |
15:21.26 | WIMPy | Siemens have combo things for both POTS and SIP, but the one I have doesn't work with multiplae phones. |
15:21.55 | WIMPy | (althouh it supports five or was it six) |
15:22.30 | FlashDeluxe | Now I`ve reinstalled dahdi, current release and when i execute dahdi_genconf i get 'Empty configuration -- no spans Empty configuration -- no spans' i guess that zaphfc misses? What can i do? |
15:24.29 | Naikrovek | anyone know a good replacement for a wrtp54g |
15:24.39 | Katty | Naikrovek: a cookie. |
15:24.47 | Katty | Naikrovek: snickerdoodles. |
15:24.56 | Naikrovek | while tasty, these do not meet my requirements |
15:25.30 | *** join/#asterisk intralanman (~lanman@va-67-76-163-226.sta.embarqhsd.net) |
15:25.32 | Katty | well you didn't list any requirements :P |
15:25.36 | Katty | just a good replacement ;P |
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15:26.49 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
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15:28.00 | Ken85 | WIMPy: yeah i want that combo hybrid thing. i'm mostly searching. i dont want to waste money. i want to be able to use lowratevoip with it. it uses G.711 so i suppose its fine. i dont believe that there would be any lags |
15:28.35 | *** join/#asterisk edguy3 (~edguy@ool-43521c56.dyn.optonline.net) |
15:28.52 | crowb4r | Hey so google is comming up a little short on a good example og asyncagi. Anyone have some links handy with good examples of using it? |
15:30.40 | *** join/#asterisk CrimsonX (ccc1490a@gateway/web/freenode/ip.204.193.73.10) |
15:31.56 | CrimsonX | why doesn't the name show up correct on inbound h323 callerid? it shows the far end IP instead |
15:33.14 | [TK]D-Fender | FlashDeluxe: Genconf is for chan_dahdi, not the core configs. /etc/dahdi/ <- |
15:33.20 | [TK]D-Fender | FlashDeluxe: system.con, etc |
15:35.03 | tzafrir_laptop | [TK]D-Fender, actually, aso /etc/dahdi/system.conf |
15:35.14 | tzafrir_laptop | But it does not run dahdi_cfg on its of |
15:35.16 | tzafrir_laptop | own |
15:36.02 | FlashDeluxe | i made reinstalled everything and i executed the patch you gave me this morning tzafrir_laptop |
15:36.25 | FlashDeluxe | but now it doesnt work |
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15:40.29 | *** join/#asterisk italorossi (~italoross@187.111.235.214) |
15:41.16 | *** join/#asterisk timeshell_atwork (~chatzilla@gw.lusi.on.ca) |
15:42.22 | carrar | snickerdoodles? |
15:42.29 | carrar | gimmie |
15:45.00 | traxx | hi. i have the following in my dialplan: exten => s,n,Dial(DAHDI/1/1234567) |
15:45.21 | traxx | i'm calling in from an external number and would like my callerid to be transferred to the new channel |
15:45.26 | traxx | any ideas how to do that ? |
15:45.51 | [TK]D-Fender | traxx: it is unless your provider or tech prevents you |
15:46.38 | zoa | you might need to tweak it a little |
15:46.52 | [TK]D-Fender | traxx: Which given you are dialing out a single channel... I'm wonding what youa re actually using... |
15:46.53 | zoa | eg. if it comes in with international prefix, but your provider will only allow national prefix etc |
15:47.09 | zoa | ah yes, you need pri or bri |
15:47.13 | traxx | going into that context from an internal number, i can manipulate the CALLERID(num), but not when calling from extern. |
15:47.16 | zoa | and bri usually doesnt allow a lot of numbers |
15:47.29 | zoa | traxx, that should not matter |
15:47.37 | zoa | unless you hardcoded the callerid somewhere |
15:47.58 | [TK]D-Fender | traxx: and I'm not seeing a call to debug or description of what you're using |
15:48.16 | carrar | You're not reading between the lines! |
15:48.21 | traxx | ok, maybe i should paste that context |
15:48.34 | [TK]D-Fender | ~pb |
15:48.35 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
15:48.36 | [TK]D-Fender | ^^^ |
15:48.56 | timeshell_atwork | I have an issue with 1.6 asterisk built in 64 bit that doesn't appear in 1.6 asterisk built in 32 bit. The issue occurs when a SIP trunk on the 64 bit build is unable to connect to it's remote host. The result is that the whole chan_sip appears to experience severe lags or hang ups. However, this doesn't occur on the server with 32 bit build. Both servers are using CentOS 5.5, 64 bit... |
15:48.57 | timeshell_atwork | ...and 32 bit respectively. |
15:50.10 | zoa | timeshell_atwork, your text is a bit confusing |
15:50.18 | timeshell_atwork | When I say server lags or hang ups, this is manifest in that all the SIP phones experience delays or inability to receive or make calls. |
15:50.27 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.166.114.dsl.dyn.forthnet.gr) |
15:50.30 | zoa | so you have 64b and 32b each trying to connect to the same 3rd server ? |
15:50.41 | timeshell_atwork | No |
15:50.53 | timeshell_atwork | I have 2 servers. a 64 bit and a 32 bit. |
15:51.21 | zoa | and the 64b calls the 32b ? |
15:51.32 | timeshell_atwork | When the internet routing is not available for any SIP trunk, chan_sip appears to become unresponsive on the 64 bit server. |
15:51.39 | zoa | aha |
15:51.43 | timeshell_atwork | However in the same situation on the 32 bit server, this doesn't happen. |
15:52.11 | zoa | i cant think for any reason for this |
15:52.31 | zoa | are you sure the rest of the config is the same ? |
15:52.32 | timeshell_atwork | They do have sip trunks to a common server, but that doesn't appear to be relevant as the issue also occurs on a trunk that loses routing that's not related to the 32 bit server. |
15:52.34 | zoa | dns etc ? |
15:53.05 | zoa | cant help im affraid |
15:53.09 | timeshell_atwork | They are both using very similar configurations, same version of asterisk. |
15:53.23 | zoa | why dont you go to 32 bit as a workaround ? |
15:53.27 | zoa | do you need >4gb memory ? |
15:53.41 | timeshell_atwork | I've thought about it. I just don't want to rebuild the server. |
15:53.42 | Qwell | I highly doubt it's an issue with bittedness |
15:53.49 | zoa | me too |
15:53.51 | timeshell_atwork | I'm open to suggestions. |
15:53.57 | zoa | i think i was the very first person to run asterisk in 64 bit |
15:54.07 | zoa | the only issue i had was md5 in iax2 |
15:54.41 | zoa | its probably still somewhere in the mantis :) |
15:54.54 | timeshell_atwork | I've tried several things to get around this issue, even creating a local server to run the internet connections and then connecting the 64 bit server to connect to it. however, even when that local link is unavailable, chan_sip still appears to have issues. |
15:55.16 | Qwell | zoa: 1804 |
15:55.27 | Qwell | or not |
15:55.40 | zoa | cant you run 32 bit on the 64 bit machine for just a second to see if it will work ? That would confirm that it is an issue with 64bit and not the config or the machine |
15:56.02 | *** join/#asterisk Jinxed- (93b128b1@gateway/web/freenode/ip.147.177.40.177) |
15:56.06 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
15:56.11 | Jinxed- | what is the default login/password for asterisk now |
15:56.14 | Jinxed- | centos |
15:56.16 | timeshell_atwork | I haven't tried compiling 32 bit on 64 bit |
15:56.30 | *** join/#asterisk philipp64|laptop (~chatzilla@63.81.41.227) |
15:56.33 | zoa | https://issues.asterisk.org/bug_view_page.php?bug_id=0001174 |
15:56.35 | timeshell_atwork | I'll look into it |
15:56.36 | zoa | this one :) |
15:56.59 | Qwell | Jinxed-: freepbx/fpbx - This is listed in the quickstart guide. You read the quickstart guide...right? |
15:57.09 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
15:57.09 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
15:57.59 | Jinxed- | Qwell: where is the quickstart guide |
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15:58.22 | zoa | qwell, do you still remember that bug that causes all asterisk machines to die in the middle of the night ? |
15:58.29 | zoa | i'd love to see some irc logs for that night :) |
15:58.33 | Qwell | http://www.asterisknow.org/AsteriskNOW-1.5-QuickStart |
15:58.33 | zoa | people trying to wake up mark |
15:58.40 | Qwell | zoa: I remember it |
15:58.44 | CrimsonX | I was on the phone, sorry |
15:58.53 | Qwell | was like right before astricon, IIRC |
15:59.04 | zoa | i dont remember that or what the date issue was |
15:59.19 | Qwell | some timestamp rolled over |
15:59.21 | zoa | aaah those memories :) |
15:59.42 | traxx | ok, here's the pastebin: http://pastebin.com/Knq2G8gd - at the top is the context, bottom the output. if i dial extension 701 from ie 702, the callerid gets set, no problem, if i dial in extension 701 from outside, i get the base number of the pbx, and not the cid i want. |
15:59.47 | zoa | things are done a bit better now :) |
16:00.00 | Qwell | there was an issue with a certain android phone. the camera focus would work for 28 days, and then not work for 28 days |
16:00.17 | CrimsonX | My setup is an h323 trunk to Avaya, and when an Avaya station call into the asterisk the CIDnumber shows up correct but the name is the IP of the Avaya instead of the caller name that is passed in the setup from the Avaya |
16:00.43 | aidinb | dahdi, oh dahdi! |
16:00.50 | zoa | yeah i remember that one |
16:01.01 | zoa | traxx, what card is that ? |
16:01.21 | zoa | try grepping all your config files for the callerid that does show up |
16:01.35 | zoa | o |
16:01.43 | traxx | zoa: from lspci: 01:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
16:01.46 | zoa | 701 from 702 fets the caller id because its all sip |
16:02.09 | zoa | is that a 10$ modem ? |
16:02.23 | zoa | or is that a bri card ? |
16:02.35 | traxx | zoa: it's bri |
16:02.50 | drmessano | ~wyd |
16:02.51 | infobot | Who's your DAHDI? |
16:02.51 | zoa | ah its the wildcard |
16:03.16 | *** part/#asterisk LemensTS (~LemensTS@adsl-70-238-154-222.dsl.stlsmo.sbcglobal.net) |
16:03.22 | traxx | i'm getting the impression that it might be a channel inheritance problem |
16:03.25 | zoa | what country are you in ? |
16:03.29 | traxx | zoa: .de |
16:03.30 | zoa | i doubt it |
16:03.40 | zoa | how many numbers can you use on the pri ? |
16:03.43 | zoa | euh bri |
16:03.44 | zoa | 8 ? |
16:04.16 | zoa | actually |
16:04.22 | zoa | so the call comes in on the bri |
16:04.28 | zoa | and gets sent to the internal number ? |
16:04.53 | zoa | your pastebin is the other way around |
16:05.02 | zoa | the way you describe will work |
16:05.04 | WIMPy | traxx: Tell your provider to enable "CLIP no screening". |
16:05.05 | zoa | the other way not so sure |
16:05.29 | zoa | WIMPy, they won't, they will only allow a few numbers on bri |
16:05.42 | zoa | i think you can only pick callerid's on pri freely |
16:05.58 | zoa | (not a tech limitation, but a provider limitation) |
16:06.11 | WIMPy | zoa: Usually they will, but they will charge. |
16:06.13 | zoa | bri is usually not for telco itnterop |
16:06.16 | traxx | like i said, calling 701 from intern redirects to the mobile number. the CID gets sent. calling 701 from outside redirects to the mobile number. i get the CID of the pbx. |
16:06.18 | zoa | not in europe afaik |
16:06.40 | zoa | ah |
16:06.43 | zoa | that makes sense |
16:07.01 | zoa | you will need to set the callerid manually |
16:07.13 | WIMPy | It only available on ptp, however, not ptmp. |
16:08.01 | zoa | so the mobile number sees 701 in the first case and in the second case your pstn callerid ? |
16:08.42 | traxx | zoa: exactly |
16:09.00 | zoa | ok |
16:09.32 | zoa | google for Set Callerid |
16:10.30 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
16:10.58 | traxx | zoa: yeah tried that (asterisk callerid redirect) but i get others with the same problem and no solution |
16:11.33 | zoa | http://www.voip-info.org/wiki/view/Setting+Callerid |
16:11.45 | zoa | will help you |
16:13.13 | traxx | tried that already. doesn't work. same effect. wouldn't know what to set the other things like name, ANI, etc. |
16:13.44 | zoa | that thing should be enough |
16:13.51 | zoa | do a debug on the bri |
16:13.56 | zoa | in both cases |
16:13.58 | zoa | and compare them |
16:14.05 | zoa | when you see a difference, come as us again |
16:14.31 | zoa | ask us i mean |
16:14.45 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-154-222.dsl.stlsmo.sbcglobal.net) |
16:14.59 | LemensTS | is libpri included in dahdi now? |
16:15.09 | Qwell | no |
16:15.23 | LemensTS | ok seen it was libpri-1.4.X ...curious thx |
16:15.34 | carrar | didn't read the notes did you :) |
16:15.46 | LemensTS | didn't even download it |
16:16.48 | *** join/#asterisk philipp64|laptop (~chatzilla@63.81.41.227) |
16:19.33 | *** join/#asterisk sylar (~sylarrrr@bzq-79-177-48-232.red.bezeqint.net) |
16:22.22 | [TK]D-Fender | traxx: I'm not seeing CALLS |
16:22.49 | traxx | [TK]D-Fender: how would i output them ? |
16:22.57 | traxx | debug ? |
16:22.59 | [TK]D-Fender | ~pb |
16:23.00 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
16:23.01 | [TK]D-Fender | ^^^^^^^^^^ |
16:23.08 | [TK]D-Fender | traxx: * CLI like everything else. |
16:24.10 | traxx | [TK]D-Fender: sorry don't understand. did you miss the calls in my pastebin ? |
16:25.33 | [TK]D-Fender | traxx: No description of what tech is used, no debug for that tech itself. |
16:26.47 | traxx | [TK]D-Fender: i 'core set debug 20' on the CLI, but the output is the same as in my pastebin |
16:27.20 | [TK]D-Fender | traxx: that is not a CHANNEL DEBUG option |
16:30.02 | *** join/#asterisk zbyniu (~zbyniu@ip-62.181.188.13.static.crowley.pl) |
16:30.35 | zbyniu | hello |
16:30.49 | carrar | HARRO |
16:31.21 | zbyniu | I have problem similar to reported here: https://issues.asterisk.org/view.php?id=17693&nbn=8 |
16:31.37 | zbyniu | but with versions 1.6.2 |
16:32.21 | zoa | i doubt it will be the bug |
16:32.26 | zoa | paste your config please |
16:32.50 | zbyniu | it looks like context is cutted somewhere in flow between chan_dahdi. and pbx.c |
16:33.08 | zbyniu | zoa: uh, it's big, but look |
16:33.36 | [TK]D-Fender | zbyniu: PASTEBIN <-------------- |
16:33.38 | [TK]D-Fender | ~pb |
16:33.39 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
16:33.41 | zbyniu | i have E1 configured with context=from-zaptel |
16:34.00 | zoa | k |
16:34.53 | zbyniu | but in log it's destroyed: VERBOSE[2738] pbx.c: -- Executing [s@tel:1] Ringing("DAHDI/2-1", "") in new stack |
16:35.38 | zbyniu | my analog lines context=from-internal are cutted to "ernal" |
16:36.06 | zoa | hmm |
16:39.10 | zoa | can you make some noops |
16:39.16 | zoa | and some ${CONTEXT} printing ? |
16:39.48 | zbyniu | sure, but where? |
16:40.13 | Carp1 | on extension 11, 11 would be my sip username? |
16:40.29 | [TK]D-Fender | Carp1: #freepbx <------------- |
16:42.00 | zbyniu | Carp1: call executing (broken name) context doesn't have proper extension no more |
16:42.05 | [TK]D-Fender | zbyniu: that bug report doesn't look like your descriptiona nd you aren't showing proper backup for it |
16:43.17 | zbyniu | [TK]D-Fender: https://issues.asterisk.org/file_download.php?file_id=26750&type=bug look here |
16:43.29 | zbyniu | Starting DAHDI/1-1 at ,s,1 still failed so falling back to context 'default' |
16:43.50 | zbyniu | it's almost the same, but context is empty here |
16:44.13 | zoa | hmm true |
16:44.37 | [TK]D-Fender | zbyniu: pastebin a SINGLE call and your configs |
16:45.03 | zbyniu | ok, w8 |
16:47.29 | ChannelZ | ok we w8 4 u lol!!1! |
16:47.29 | *** join/#asterisk Asinus1223 (~MVCoon@adsl-190-81-134.asm.bellsouth.net) |
16:47.57 | Asinus1223 | Hello everyone |
16:48.14 | Asinus1223 | Is there a place to look up stoopid newbie questions |
16:48.23 | [TK]D-Fender | Asinus1223: bash.org |
16:48.26 | ChannelZ | The Google |
16:48.56 | pabelanger | zbyniu: pb a full debug log |
16:49.12 | pabelanger | ~collectdebug |
16:49.13 | infobot | rumour has it, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
16:49.16 | pabelanger | zbyniu: ^^ |
16:49.30 | Qwell | hmm |
16:49.40 | Qwell | ~collectdebug @ pabelanger |
16:49.45 | Qwell | lame |
16:49.52 | Qwell | ~collectdebug > pabelanger |
16:49.55 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
16:49.57 | Qwell | oh well |
16:50.31 | drmessano | Can't he just tell you how it SOUNDS? |
16:50.35 | drmessano | or LOOKS? |
16:50.42 | drmessano | Why all this "DEBUG" crapola stuff |
16:50.51 | drmessano | ????11!!!1!!!??? |
16:51.02 | zoa | he already told us how it smells |
16:51.23 | drmessano | Then why hasn't some developer fixed it yet? |
16:51.56 | pabelanger | o.0 |
16:52.03 | Asinus1223 | I have a TMD400 I can't get a dial tone off of the FXS port, and I was just wondering if I have the wires plugged in right. |
16:52.16 | [TK]D-Fender | drmessano: You'd love this one... a new girl in marketing called me over to look at her phone problem. She says whenever she was calling this one person she'd get a EHN EHN EHN EHN sound. I looked at the phone for a second and told her "Every heard of a BUSY SIGNAL?" |
16:52.19 | Asinus1223 | Should I plug the FXO prot prior to the demarq? |
16:52.33 | zoa | Asinus1223: if its an fxs, try to connect a normal phone |
16:52.42 | [TK]D-Fender | Asinus1223: did you connect the molex? |
16:52.53 | Asinus1223 | should it give a dial tone without any fussing with the config files? |
16:52.59 | drmessano | LOL |
16:53.07 | Asinus1223 | yeah, the molex was the fiurst thing I did |
16:53.42 | Asinus1223 | and my head is buzzing with channels and extensions and contextx |
16:53.50 | zoa | we need a new person to say NEXT! |
16:53.58 | Carp1 | NEXT |
16:54.02 | Qwell | ~NEXT |
16:54.03 | infobot | somebody said next was NEXT! |
16:54.19 | carrar | PREVIOUS |
16:54.24 | Carp1 | ~infobot |
16:54.25 | infobot | carp1, i love abuse, feed me!, or whack, yo |
16:54.29 | Asinus1223 | CURRENT |
16:55.00 | carrar | better yet |
16:55.04 | carrar | PENULTIMATE |
16:55.07 | [TK]D-Fender | Asinus1223: Did you initialize DAHDI? Confirmed the channels were configured and ack'd by * CLI? |
16:56.28 | zoa | do you have fxs modules on that card ? |
16:56.41 | zoa | or am i completely mistaken ? :) |
16:56.57 | Asinus1223 | I don't have the `dahdi show channel `command available |
16:57.03 | zoa | oh oh |
16:57.16 | Asinus1223 | and yes, I have one green FXS module and one orange FXO module |
16:57.28 | [TK]D-Fender | Asinus1223: No DAHDI running in * = no dialtone |
16:57.37 | [TK]D-Fender | Asinus1223: Fix |
16:57.38 | Asinus1223 | ah yes. |
16:57.49 | Asinus1223 | lsmod shows dahdi runninng |
16:58.07 | zbyniu | http://pastebin.com/s0hAqndG |
16:58.07 | Asinus1223 | I guess it isn;t runing "in asterisk" |
16:58.11 | [TK]D-Fender | Asinus1223: Irrelevant. thats like having a working motor... that isn't installed in the car |
16:59.53 | carrar | Would the engine light come on without a engine? |
17:00.07 | Naikrovek | if the computer is still there, yes |
17:00.11 | Asinus1223 | OK is there some doc that shows how to have asterisk access the dahdi dirvers? I thought this was taken care of by the `dahdi_genconfig` command |
17:00.30 | [TK]D-Fender | Asinus1223: did you look at the result? |
17:00.39 | [TK]D-Fender | Asinus1223: Did you check that chan_dahdi.so is loaded? |
17:00.46 | [TK]D-Fender | Asinus1223: Did you try to load it manually? |
17:00.53 | [TK]D-Fender | Asinus1223: Why is the sky blue? |
17:01.09 | [TK]D-Fender | Asinus1223: What is the average air-speed velocity of an unladen swallow? |
17:01.19 | [TK]D-Fender | Asinus1223: Who shot J.R.? |
17:01.38 | Asinus1223 | Yeah, I looked at the result. It produced a couple of files; /etc/dahdi/system.conf and the other /etc/asterisk |
17:01.51 | [TK]D-Fender | Asinus1223: And what did YOU produce? :) |
17:02.00 | drmessano | It was Kristen, Sue Ellen's sister |
17:02.06 | carrar | 24 miles an hour |
17:02.09 | drmessano | Bitch.. hated her |
17:02.11 | [TK]D-Fender | drmessano: NO SPOILERS BITCH! |
17:02.16 | Asinus1223 | ozone particles |
17:02.16 | drmessano | HA |
17:02.35 | drmessano | [TK]D-Fender, I JUST RUINED THE 1981 TV SEASON FOR EVERYONE |
17:03.01 | zbyniu | pabelanger: ok, i'll try to cut configs to some minimal set and pb it if http://pastebin.com/s0hAqndG is not enough |
17:05.19 | *** join/#asterisk andresm (~andresm@ubuntu/member/andresmujica) |
17:05.41 | [TK]D-Fender | zbyniu: Poor. unload the module. Load it. Show us the channel definitions. Retes |
17:05.45 | [TK]D-Fender | retest* |
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17:10.13 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
17:10.13 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
17:11.25 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
17:16.06 | zbyniu | [TK]D-Fender: chan_dahdi? nothing special: http://pastebin.com/xdZSczYE |
17:16.46 | [TK]D-Fender | [Jul 29 19:13:22] WARNING[3257]: chan_dahdi.c:17018 process_dahdi: Ignoring any changes to 'hassip' (on reload) at line 35. |
17:16.48 | [TK]D-Fender | ^^ |
17:16.49 | [TK]D-Fender | PARDON? |
17:17.00 | [TK]D-Fender | zbyniu: Go look in USERS.CONF while you're at it |
17:17.02 | Qwell | asterisk-gui |
17:17.12 | [TK]D-Fender | ##$#$*$*$%#%(@&#^)!@#^#_&^@#$_*#^($(_#^&()^&*)@#^&^@%#$ |
17:18.13 | zbyniu | [TK]D-Fender: and? hassip = yes |
17:20.56 | Naikrovek | [TK]D-Fender: watch this right up to the very last second, and laugh a bit: http://www.funnyordie.com/videos/ed36fa1ab6/between-two-ferns-with-zach-galifianakis-steve-carell |
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17:45.03 | zbyniu | hmm, myDebugLog generated via HOWTO_collect_debug_information.txt instruction has 311MB, are you sure I should paste it somewhere? :) |
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17:48.36 | tzafrir_laptop | zbyniu, the core file itself? no |
17:48.45 | tzafrir_laptop | the trace from the gdb - yes |
17:50.05 | tzafrir_laptop | zbyniu, WARNING[3074] chan_dahdi.c: Unable to enable echo cancellation on channel 10 (No such device) |
17:50.30 | tzafrir_laptop | Could you please pastebin the output of: lsdahdi ? |
17:50.36 | zbyniu | tzafrir_laptop: i can enable it if you want, doesn't matter |
17:51.28 | tzafrir_laptop | zbyniu, you seem to have definitions in users.conf as well |
17:51.59 | tzafrir_laptop | If you rely on chan_dahdi.conf and extensions.conf, I would suggest that you remove / move aside users.conf |
17:52.05 | stix | What does this reply mean? SIP/2.0 488 Not acceptable here |
17:52.17 | tzafrir_laptop | Or at least the parts of it that define DAHDI channels |
17:52.37 | anonymouz666 | stix: could be a codec offer that result this |
17:52.45 | stix | hmm okay |
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17:53.50 | zbyniu | tzafrir_laptop: http://pastebin.com/wxphDqjC - users.conf |
17:54.13 | stix | might be what it's saying here: Capabilities: us - 0x2 (gsm), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) |
17:54.27 | anonymouz666 | combined NOTHING |
17:54.49 | stix | trying with alaw |
17:55.01 | tzafrir_laptop | zbyniu, if that's all, it's harmless |
17:55.05 | stix | wee! :) |
17:55.06 | b11d` | had a compiler warning (in 1.8-b2) about 'strudpa' in sig_pri.c -- modified strupda to ast_strdupa and that fixed the problem.. chan_dahdi loads now and so does pri. |
17:55.16 | b11d` | dunno if I should submit a bug report for such a small fix.. |
17:55.28 | b11d` | it was causing chan_dahdi to not load in FreeBSD 8.1-amd64 anyway |
17:55.29 | tzafrir_laptop | b11d`, please do |
17:55.36 | tzafrir_laptop | On what platform is this? |
17:55.49 | b11d` | FreeBSD 8.1-amd64 |
17:56.05 | zbyniu | tzafrir_laptop: yes, that's all |
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18:04.26 | b11d` | bug report submitted |
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18:12.58 | malcolmd | b11d`: and a "real" diff attached. thank you :) |
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18:14.37 | *** mode/#asterisk [+o Qwell] by ChanServ |
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18:16.48 | b11d` | no, thank you :) |
18:17.55 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
18:17.55 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
18:20.36 | Katty | ih |
18:20.38 | Katty | hi |
18:23.54 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
18:24.18 | leifmadsen | Katty: ohai! |
18:24.26 | wcselby | o/ |
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18:27.10 | b11d` | yay I contributed :) |
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18:29.44 | Katty | hugs leifmadsen |
18:29.51 | Katty | hugs wcselby :> o |
18:30.23 | Katty | so what fragrance do you guys wear. |
18:30.27 | Katty | what's your favorite |
18:30.33 | leifmadsen | wow... my router has decided it wants to be chinese... |
18:30.33 | wcselby | whatever soap my wife buys |
18:30.42 | leifmadsen | I wonder how the heck I switched the language on it.... |
18:30.57 | malcolmd | leifmadsen: how do you know that someone in china didn't decide that for you? |
18:31.16 | leifmadsen | malcolmd: I don't? :) |
18:31.37 | zbyniu | [TK]D-Fender,tzafrir_laptop,pabelanger: https://issues.asterisk.org/view.php?id=17753 |
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18:37.16 | tzafrir_laptop | zbyniu, hmm.... can you pastebin the output of: dahdi show channel 1 ? |
18:38.05 | wcselby | i've got a client that their dahdi pseudo device stops working with some frequency, preventing them from being able to access any meetme bridges. they are still able to make and receive calls over their T1 lines. where should I begin looking? I'm getting a cli trace ready |
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18:39.45 | tzafrir_laptop | wcselby, what does dahdi_test show at the time of such a hang? |
18:40.11 | tzafrir_laptop | Do they have more than one T1 line on that box? |
18:40.44 | evilbit | if I'm in a context called internal and I want to dial a extension in a context of local-call do I need to include local-call in the context of internal? |
18:40.48 | wcselby | tzafrir_laptop - yes, 3 lines going into a TE420p, and a 4th line that runs into a brooktrout board on a separate hylafax server |
18:41.00 | wcselby | it's a te420, don't remember the last number |
18:41.28 | wcselby | 4 port T1 PCI-e card with echo cancel board |
18:42.25 | [TK]D-Fender | evilbit: Yes |
18:43.04 | evilbit | hmm, actually I have that set... I am unable to dial out via my iax provider. They are saying it's because I'm not setting the CID but AFAIK I am |
18:43.36 | [TK]D-Fender | evilbit: If you say so |
18:43.47 | evilbit | Set(CALLERID(number)=${OUTCID}) should set it right? If in sip.conf for my user I have SetVar=OUTCID=<my did here> |
18:44.59 | zbyniu | tzafrir_laptop: http://pastebin.com/psVUEWwK |
18:46.26 | wcselby | tzafrir_laptop - http://pastebin.com/z5TBpAqD requested output, plus cli from the failed meetme attempts |
18:47.40 | [TK]D-Fender | evilbit: You might want to try actually looking at the call |
18:48.13 | wcselby | I can bandaid the issue by stopping asterisk, then dahdi, the starting dahdi, then asterisk. 30 seconds of no calls though. we usually have about 15-20 concurrent calls going during the day though |
18:48.15 | evilbit | [TK]D-Fender: well, I have and I can see it go out and * says it's accepted by the remote end but then nothing happens :-( |
18:49.06 | [TK]D-Fender | evilbit: And you are showing us nothing. Expect our ability to assist to be proportionate |
18:49.17 | evilbit | sorry... 1 sec |
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18:50.35 | tzafrir_laptop | zbyniu, there's something a bit unusual that you do: you don't provide an actual extension, and rely on falling back to 's' |
18:50.56 | tzafrir_laptop | What happens if you also provide a catch-all extension '_X.' ? |
18:50.57 | evilbit | it just stops here: http://pastebin.com/Ly48QQq4 |
18:52.08 | tzafrir_laptop | zbyniu, btw: nice catch |
18:54.43 | [TK]D-Fender | evilbit: Who are you calling? |
18:55.16 | evilbit | well, right now I'm trying to call my cellphone (but it's just for testing... I can't call any number) |
18:55.49 | evilbit | it just stops at the "Format for call is ulaw" |
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18:57.29 | evilbit | but if I dial another sip number then the CID that I set doesn't show up so I'm wondering if they are right and I have something wrong |
18:58.25 | pabelanger | zbyniu: Did you pastebin 'dahdi show channel 1' ? |
18:59.29 | tzafrir_laptop | pabelanger, he did. THe context shows there properly: "from-zaptel" |
19:00.19 | evilbit | hmm, ok... so if I remove the SetVar from sip.conf then I can dial out |
19:00.57 | [TK]D-Fender | evilbit: - Executing [12158872972@ip-solutions-internal:1] Set("SIP/hhoffman-desktop-0000000d", "CALLERID(number)=") in new stack <--- shouldn't have QUOETS here. |
19:01.02 | [TK]D-Fender | EQUOTES* |
19:01.04 | [TK]D-Fender | sjdfhjsdfgsdf |
19:01.06 | [TK]D-Fender | gah |
19:01.54 | evilbit | hmm, ok |
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19:04.42 | evilbit | here are the relevant bits from extensions.conf: http://pastebin.com/JSHe7hQN |
19:05.33 | zbyniu | pabelanger: http://pastebin.com/psVUEWwK - dahdi show channel 1 |
19:06.05 | pabelanger | zbyniu: working on a patch for you to try |
19:06.14 | zbyniu | tzafrir_laptop: if i remove ext tel it will go to 'default' |
19:06.52 | pgrace | where does oej's srtp patch live nowadays? |
19:07.05 | leifmadsen | pgrace: in the 1.8 branch and trunk |
19:07.06 | pgrace | the http://www.e164.org/wiki/AsteriskSRTP link is wrong. |
19:07.13 | leifmadsen | pgrace: it's been merged already |
19:07.14 | pgrace | Is there a 1.6 patch still? |
19:07.19 | tzafrir_laptop | zbyniu, hmm... actually what I wrote is pointless. It found that extension in context 'tel' anyway, so we have to worry indeed about the context first |
19:07.23 | leifmadsen | pgrace: if there is, it's very old and broken |
19:07.26 | pgrace | ok |
19:07.27 | zbyniu | pabelanger: great, i've builder and pbx ready to test :) |
19:07.31 | evilbit | aha! It's because Wireless Caller was in quotes |
19:07.37 | pgrace | leifmadsen: 1.8 is still considered devel right? |
19:07.37 | leifmadsen | pgrace: it'll be on https://issues.asterisk.org as a closed issue |
19:07.49 | leifmadsen | pgrace: the fact we have 1.8.0-beta2.... yes :) |
19:07.55 | pgrace | leifmadsen: :) |
19:08.00 | pgrace | drats |
19:08.03 | leifmadsen | please help test |
19:08.39 | pgrace | yeah, sounds like I gotta toss up a test vm |
19:08.42 | pabelanger | zbyniu: http://asterisk.pastebin.com/qkdrKFAg |
19:08.47 | pabelanger | zbyniu: untested |
19:09.00 | pgrace | leifmadsen: working on OCS and asterisk compatibility when not using a mediation server, it's... interesting. |
19:09.09 | pgrace | pretty sure my problem is SRTP. |
19:09.29 | wcselby | anyone have any thoughts on why my pseudo dahdi device isn't working - http://pastebin.com/z5TBpAqD |
19:09.39 | wcselby | or have any other troubleshooting steps I can do other than restart dahdi |
19:09.57 | pabelanger | wcselby: permissions? |
19:10.06 | LemensTS | ERROR[2097]: chan_dahdi.c:10180 mkintf: Signalling requested on channel 1 is ISDN PRI but line is in FXO Kewlstart signalling <---I dont have fxoks set anywhere in /etc/dahdi/system.conf nor /etc/asterisk/chan_dahdi.conf |
19:10.23 | wcselby | pabelanger - same perms on /dev/dahdi/pseudo as any of the other /dev/dahdi/ devices |
19:10.42 | tzafrir_laptop | LemensTS, what version of Asterisk is it? |
19:10.57 | LemensTS | 1.6.2.10 |
19:11.08 | tzafrir_laptop | LemensTS, and it's actually fxo_ks in chan_dahdi.conf |
19:11.27 | tzafrir_laptop | (or users.conf) |
19:12.36 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
19:13.59 | wcselby | dahdi version 2.2.1.2, vpm450 hardware echo can |
19:14.15 | wcselby | asterisk 1.4.33.1 |
19:16.50 | LemensTS | tzafrir_laptop: http://pastebin.com/t40awzE9 here is pastebin of chan_dahdi.conf |
19:16.55 | malcolmd | wcselby: is it something that's resolved by restarting asterisk or only by restarting (unloading and reloading the kernel modules) dahdi? |
19:17.12 | malcolmd | wcselby: happens always, regularly, periodically (fixed or variable)? |
19:17.53 | wcselby | malcolmd - only resolved by restarting dahdi |
19:18.10 | wcselby | happens more frequently than it used to (vague, I know) |
19:18.18 | wcselby | i first noticed this when I installed oslec |
19:18.24 | wcselby | then I removed oslec and installed the hardware board |
19:18.30 | LemensTS | here it is with system.conf at the bottom http://pastebin.com/Gj9WS2d1 |
19:18.30 | wcselby | and didn't see it for a few months |
19:18.39 | wcselby | now it's happened three days in a row |
19:18.46 | wcselby | each time I had to restart dahdi |
19:19.03 | malcolmd | wcselby: this sounds like a "fun" one for our support team, i'm afraid. http://www.digium.com/support |
19:19.12 | wcselby | it happened once last week and maybe once the week before |
19:19.15 | wcselby | yeah |
19:19.33 | wcselby | i was afraid of that |
19:19.42 | wcselby | i need to get the serial number of the card to get support, yes? |
19:20.04 | malcolmd | yup. |
19:20.16 | tzafrir_laptop | LemensTS, the 'group' setting at the end of chan_dahdi.conf has no effect, BTW |
19:20.21 | tzafrir_laptop | anything in users.conf? |
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19:22.08 | LemensTS | tzafrir: added users.conf and output of trying to load the chan_dadhi.so from cli output http://pastebin.com/XQN8aMTz |
19:22.12 | LemensTS | *at bottom |
19:22.42 | bdheeman | hello |
19:23.17 | bdheeman | is this http://svn.digium.com/svn/asterisk/trunk and 1.8.0 head? |
19:23.39 | Qwell | bdheeman: what? |
19:23.55 | *** join/#asterisk nix8n82 (~nate@63.162.27.14) |
19:24.05 | malcolmd | bdheeman: 1.8 branch is http://svn.digium.com/svn/asterisk/branches/1.8 |
19:24.15 | bdheeman | ok |
19:24.37 | tzafrir_laptop | LemensTS, I read that message backwards |
19:24.42 | bdheeman | and what's the above said trunk is? |
19:24.46 | tzafrir_laptop | try running dahdi_cfg again |
19:26.28 | malcolmd | bdheeman: trunk is http://svn.digium.com/svn/asterisk/trunk |
19:27.03 | LemensTS | that gets me: DAHDI_SPANCONFIG failed on span 1: Invalid argument (22) <--does 22 mean line number or is it an error code? |
19:28.07 | bdheeman | malcolmd: which one you recommend should I try with a2billing 1.7.1~svn? |
19:29.13 | bdheeman | malcolmd: for experimentation and testing |
19:29.51 | malcolmd | bdheeman: best to shy away from trunk as that's the source of all new development, and thus it's considered completely unstable and isn't guaranteed to compile. 1.8's a better place to go. currently, we're in the beta process for 1.8 (we're at beta2 as of a few days ago). so, what you check out of the 1.8 branch will eventually become 1.8.0 for general availability. |
19:30.33 | bdheeman | malcolmd: thanks |
19:31.34 | zbyniu | LemensTS: error code |
19:32.11 | *** join/#asterisk bakermd (~bakermd@66.117.118.34) |
19:32.15 | malcolmd | bdheeman: hrm..i'm looking at the a2billing page and it says that 1.7.1 stable was released in may of 2010. when they released that, the latest asterisk was probably 1.6.2. so, they may not yet be working on 1.8. i'm looking at their documentation page (http://www.asterisk2billing.org/cgi-bin/trac.cgi/wiki/Documentation) and it says Asterisk 1.4.0+. so, they may not even work with asterisk 1.6 (which was before asterisk 1.8). |
19:32.38 | malcolmd | so, if that's the case, then you can check out from http://svn.digium.com/svn/asterisk/branches/1.4 |
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19:33.22 | *** join/#asterisk davesullivan (44415202@gateway/web/freenode/ip.68.65.82.2) |
19:33.30 | bakermd | I put app_fax.so in my asterisk modules folder and restarted asterisk, however when I do `core show applications` I do not see fax listed... starting asterisk in non-forking mode does not display any errors - any ideas? |
19:33.41 | malcolmd | bdheeman: if you're planning to build something permanent, you might want to review the Asterisk Version release and EOL schedule here (http://www.asterisk.org/asterisk-versions) |
19:33.45 | bdheeman | malcolmd: ok, I checked out http://svn.digium.com/svn/asterisk/trunk in around 2008 |
19:34.03 | pabelanger | bakermd: *CLI> module load app_fax.so |
19:34.20 | zbyniu | pabelanger: nothing changed with your patch |
19:35.10 | bdheeman | malcolmd: thanks for the link, reviewing it right now |
19:35.42 | bakermd | pabelanger: Unable to load module app_fax.so || Command 'module load app_fax.so' failed. -- any idea how I can get additional info out of it? |
19:36.41 | pabelanger | zbyniu: Then you have a different issue then what I was thinking |
19:37.05 | bakermd | pabelanger: I see the full logging now - thanks |
19:37.21 | bdheeman | malcolmd: from that page, seems better I try 1.8 branch |
19:38.43 | malcolmd | bdheeman: as far as asterisk goes, yes. i just don't know the state of a2billing development related to asterisk 1.8 support |
19:38.54 | pabelanger | zbyniu: Did you update Asterisk recently? Did this work previously? |
19:39.13 | zbyniu | pabelanger: can you explain me in 2 words how incoming connection goes to extension resolve code? |
19:39.20 | bakermd | Any thoughts on error "undefined symbol: ast_register_application" |
19:39.26 | zbyniu | from chan_dahdi.c: -- Accepting call from <- here is ok |
19:39.52 | zbyniu | to: pbx.c: == Starting DAHDI/2-1 at tel,0,1 failed so falling back to exten 's' |
19:40.16 | zbyniu | pabelanger: i have oooold asterisk now, and working on new |
19:40.31 | zbyniu | tried asterisk 1.6.2.6 and .10 |
19:40.44 | zbyniu | dahdi 2.2.1 and 2.3.0 |
19:40.45 | bdheeman | malcolmd: NP, I'll test and revert back to 1.6 or even 1.4 release if needed |
19:41.07 | zbyniu | wanpipe 3.5.10 and .14... :) |
19:42.56 | pabelanger | zbyniu: did you pb you chan_dahdi.conf? |
19:43.23 | malcolmd | bdheeman: cool :) |
19:44.18 | bdheeman | malcolmd: :) |
19:45.05 | zbyniu | pabelanger: one part here: http://pastebin.com/s0hAqndG |
19:45.05 | Qwell | bakermd: You aren't loading the module in the correct version of Asterisk. |
19:45.14 | bakermd | Qwell: Aah - thanks |
19:47.36 | *** join/#asterisk MatBoy (~MatBoy@wiljewelwetenhe.xs4all.nl) |
19:49.44 | davesullivan | Anyone have any experience running asterisk on ec2? Seems like a physical machine works a bit better? |
19:50.00 | Qwell | davesullivan: no reason it shouldn't work |
19:50.06 | nightwalk | Finally caught a capture of my asterisk installation acting up, though the output is sparse despite having set core debug to 99: http://pastebin.com/8NXmpiEs As near as I can tell, this dmesg output goes with it (but would seem to be useless): http://pastebin.com/k0K7raXn |
19:50.58 | malcolmd | b11d`: and...committed, issue closed. thanks again |
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19:52.15 | nightwalk | Basically, what happened is a user called out to the POTS from an extension off of a TDM410, and was disconnected for reasons unknown ~1 minute later. The user hung up the phone, and it started ringing. No one was there when they answered it. |
19:52.58 | davesullivan | Qwell: yeah, i've got one up and running, but I keep reading things about it being a bad idea due to some kind of timer thing |
19:53.31 | davesullivan | Seems to work well enough, but I get garbled audio issues occasionally and haven't tested it fully loaded w/ 20 calls going at once yet |
19:53.59 | nightwalk | I'm sure this ties in with the mysterious dropping call problem I've been trying to track down and fix, though the ghost ringing is new. Anyone have any ideas? I don't see anything in the trace that I didn't already know :/ |
19:54.18 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
19:55.18 | nightwalk | Asterisk 1.6.2.5-0ubuntu1, with DAHDI 2.2.1, btw |
19:57.39 | malcolmd | nightwalk: hit our support department up on this one, please; i think they're best-suited to help troubleshoot and resolve. http://www.digium.com/support |
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19:58.50 | nightwalk | malcolmd: Any particular reason you say that? |
19:59.32 | citywok | when a call is blind transferred to another person, the call has the same callid, and the same recording file. the problem i have is the first person that answered the call transfers, and a CDR is generated. i process the CDR, encode the WAV file and delete it. i have no way of knowing this is a transfer and the file is still being written to. is there any solution for this? |
20:00.20 | hardwire | hmm. |
20:00.28 | nightwalk | Also, not sure Digium will support this card. It *does* appear to be a full-fledged Digium card ('Digium' *is* printed on it, and it *isn't* from china), but I didn't get it through them. |
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20:13.20 | citywok | nightwalk: considering how awful digium support is, even if you do get a response it could take weeks. |
20:13.31 | b11d` | no prob malcolmd... glad I could help for once :) |
20:13.40 | wcselby | lol, i've not had any problems with digium |
20:14.03 | *** part/#asterisk bdheeman (~bdheeman@bdheeman-2-pt.tunnel.tserv20.hkg1.ipv6.he.net) |
20:14.17 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
20:14.43 | nightwalk | citywok: No support from them anyway. According to malcolmd, my card is trademark-infringing knock-off. |
20:15.02 | nightwalk | ...from canada |
20:15.42 | wcselby | damn canadians |
20:15.49 | wcselby | america's hat strikes again! |
20:15.54 | wcselby | :P |
20:16.39 | citywok | wcselby: 72 hour response times and they respond in the middle of the night b/c their email support team for the cheap support is done at night |
20:16.52 | *** join/#asterisk Benwa (~Benwa@unaffiliated/benwa) |
20:17.13 | citywok | oh, and when their device has a crippling bug they admit in october, they still have not released a fix that keeps their PBX appliacnce from hard locking every week because they are working on adding new features. |
20:17.20 | nightwalk | Actually, I knew what I was getting. I don't really care if it's Digium or not, so long as it works. Digiums are far too expensive to use in proof-of-concept implementations |
20:17.21 | wcselby | citywok - called in at 7 am EST, answered on the second ring, guy helped me as much as he could (it was a hardware issue) |
20:17.42 | citywok | thanks, i dont want any freaking features, i want you to make it so my AA50 doesn't go read-only on the CF card every few days lighting the thing on fire |
20:17.51 | citywok | oh, i dont get phone support b/c the thing only came with email support |
20:17.56 | pabelanger | nightwalk: what company in Canada? |
20:18.05 | nightwalk | pabelanger: nicherons |
20:18.12 | wcselby | citywok - well, we've obviously had very different reactions |
20:18.17 | wcselby | erm, not reactions |
20:18.23 | wcselby | uh, experiences :) |
20:18.33 | citywok | yea. you are lucky for getting phone support :) |
20:21.04 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
20:22.35 | nightwalk | Funny thing is, with all of the 'compatible' devices out there, Digium's prices seem to stay unreasonably high. I mean, I can get a whole populated tdm and still have money left over for what I'd pay for a bare Digium tdm |
20:23.11 | citywok | i have a pair of the 24 port cards loaded on a shelf |
20:23.25 | citywok | bought right off of digium's website like 2 years ago |
20:23.38 | citywok | (if you are looking to buy one or two that is) |
20:23.45 | Qwell | nightwalk: Ever heard the expression "You get what you pay for"? |
20:23.59 | nightwalk | qwell: I was waiting for that to come up, actually :) |
20:24.14 | Qwell | well, if you're buying cheap garbage, you really shouldn't expect it to act otherwise.. |
20:24.27 | Qwell | however, having said that. |
20:24.35 | citywok | we used a couple knockoff T1 cards that worked fine, but no knockoffa nalog stuff |
20:24.42 | Qwell | Price similar hardware from other PBX vendors . You'll cry. |
20:24.48 | nightwalk | Like other brands, Digium advocates always bring that up. Problem is, I don't have any experience with Digium, so they may well be as bad as the next guy. |
20:25.10 | [TK]D-Fender | [16:00]<nightwalk>Also, not sure Digium will support this card. It *does* appear to be a full-fledged Digium card ('Digium' *is* printed on it, and it *isn't* from china), but I didn't get it through them. <-- I've seen these fakes on eBay before. |
20:25.14 | nightwalk | Paying less means opening myself up to that much less of a rip-off |
20:25.18 | [TK]D-Fender | also reported them to Digium |
20:25.20 | citywok | Qwell: no kidding. it's still 1/100th the cost per port of an off the shelf phone system |
20:25.23 | Qwell | nightwalk: countless people do. you never see people praising the crap hardware. |
20:25.44 | hardwire | any good way to completely stop media progress? |
20:25.48 | hardwire | err progress media? |
20:25.53 | Qwell | hardwire: ctrl-c :D |
20:25.54 | hardwire | I have it prohibited to the nth degree |
20:25.57 | hardwire | Qwell: nono |
20:25.58 | hardwire | nonono |
20:26.15 | nightwalk | [TK]D-Fender: Yes, malcolmd already established that ;) |
20:27.39 | nightwalk | Anyway, I'm thinking the issue might be with the hardware the card is in. Problem is, I don't see anything in the traces I posted to support that. Any way to confirm without transplanting the card? |
20:29.48 | *** join/#asterisk IamTrying (~IamTrying@94-227-2-47.access.telenet.be) |
20:34.45 | *** join/#asterisk Alagar (~Administr@122.164.32.124) |
20:35.22 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
20:35.48 | nny | I use the fake digium cards, but they say sangoma on them. Work better too! |
20:35.52 | nny | :D |
20:36.34 | b11d` | im not buying another sangoma card in my life |
20:36.43 | b11d` | i liked them, but zero support lately... zero. |
20:36.45 | nny | lol that's how I feel about digium's cards |
20:36.55 | nny | really? odd. I can usually call and hit a support tech almost right away |
20:37.03 | b11d` | yeah for like two weeks, calls and emails.. no reply. |
20:37.13 | nny | my first t1 digium card with hardware ec sucked hard. Had to put a telelabs EC in front of it |
20:37.16 | b11d` | only bought six a104d cards |
20:37.48 | b11d` | im buying my first digium card next week |
20:38.54 | b11d` | after spending almost $10k on sangoma cards, I expect at least some kind of reply from their support people.. even a "we dont know, fuck off!" would be fine haha |
20:40.39 | Kobaz | sangoma support is really great |
20:40.43 | b11d` | it used to be |
20:41.07 | b11d` | i've had good experiences with them in the past as well, just not lately |
20:41.15 | *** join/#asterisk cmendes0101 (~nn@pool-96-251-59-245.lsanca.fios.verizon.net) |
20:41.51 | Kobaz | what's the current issue? |
20:42.41 | b11d` | im experiencing a problem getting a D-channel operational on a fractional PRI, where only channels 1-11 are available, with D on channel 24. My simple question to them is, whats the best way to configure wanrouter to support this setup? |
20:43.07 | b11d` | the card works flawlessly on a full 24-channel PRI in another location.. |
20:43.48 | b11d` | and I've been over it a dozen times w/ the telco.. they say the PRI is good and in working order |
20:44.07 | Kobaz | have you checked at the smartjack |
20:44.33 | b11d` | yes, as I said, I've been over it w/ the telco a dozen times |
20:44.55 | Kobaz | and nothing interesting from the sangoma logs? |
20:45.08 | b11d` | granted, all I can do is accept their statement of "its working fine" -- not much I can do to investigate myself. |
20:45.23 | Kobaz | yeah... other than dialing into the smartjack |
20:45.27 | wcselby | what is the differrence between -nocana and -opteron with regard to the hpec binary? |
20:45.35 | b11d` | which I cant do |
20:46.07 | b11d` | no, not much interesting in the logs... i've tried manually setting the DCHAN and the AVAIL_CHAN options in wanpipeX.conf but to no avail.. |
20:46.14 | b11d` | says the D channel is up, but its not.. |
20:46.42 | b11d` | so, all im looking for from them is verification that im not doing something completely stupid and that my config SHOULD work, so I can go BACK to the telco and say its their end :) |
20:46.43 | nightwalk | I believe Sangoma is the one the who makes the fax card we use. They were the 'economy model' cards, but still cost like 10k. The 'replacement' cards were $20k+. Given that, and the fact that they don't seem to want to play nicely with linux, I'd imagine they probably won't be getting any more of our business. |
20:47.55 | Kobaz | i've had no problems with sangoma on linux other than dahdi problems which isn't sangoma's fault |
20:48.13 | nny | hmm guess opinions vary, i have had nothing but good luck with sangoma cards, and nothing but nightmares with digium ones |
20:48.19 | Kobaz | libpri/dahdi has just been really crappy for me lately... i've switched to external gateways |
20:48.20 | b11d` | and, two weeks of emailing them and leaving messages w/ support, and NOT A SINGLE REPLY IN ANY WAY. |
20:48.35 | nny | yeah that sucks b11d, maybe you need to contact someone higher up |
20:48.38 | Kobaz | give them a call |
20:48.41 | Kobaz | they do phone support |
20:48.43 | b11d` | I have been calling |
20:48.49 | b11d` | I have been emailing |
20:48.59 | b11d` | as I said... my Digium card order goes in next week... |
20:49.04 | b11d` | they've lost me |
20:49.15 | nny | heh well. Don't be suprised if digium isn't much better |
20:49.16 | b11d` | I'll try my luck w/ Digium |
20:49.20 | Kobaz | heh |
20:49.24 | b11d` | well at least I can come here and get help :) |
20:49.27 | b11d` | no #sangoma |
20:49.28 | b11d` | :P |
20:49.43 | nny | i have replaced more FXO modules than I can count, i got a box of them at the office |
20:49.50 | Kobaz | yeah |
20:50.07 | Kobaz | at the last place i worked.. fxos would fry pretty often |
20:50.15 | b11d` | poorly grounded wiring? |
20:50.27 | NEEDINGHELP123 | Hi Guys |
20:50.30 | NEEDINGHELP123 | hows it going? |
20:50.34 | b11d` | great, friend! |
20:50.41 | NEEDINGHELP123 | good news |
20:50.45 | citywok | i'm going out on a limb here, and guessing you need help? |
20:50.47 | Kobaz | no idea... went with sangoma and the modules stopped frying |
20:50.55 | NEEDINGHELP123 | no man |
20:50.55 | nny | dunno, it's the telco 66 block --> card |
20:50.58 | NEEDINGHELP123 | everythign good |
20:51.05 | NEEDINGHELP123 | no help needed today |
20:51.05 | nny | lol |
20:51.09 | cmendes0101 | Trying to add a register => into sip.conf but im getting this error: [Jul 29 21:50:02] WARNING[4266]: chan_sip.c:2921 __sip_xmit: sip_xmit of 0x7f025445e9f0 (len 407) to 206.15.156.221:5060 returned -1: Address family not supported by protocol |
20:51.10 | NEEDINGHELP123 | maybe tomorrow ;) |
20:51.27 | citywok | nny, we had a pair of 24 port cards that never ate a single module, all punched in to 66 blocks and SLC's in our old phone system |
20:51.43 | citywok | they were the digium tdm2400p's i think |
20:51.53 | nny | citywok: yeah guess it all depends on the telco wiring |
20:51.58 | NEEDINGHELP123 | I may need to employ an asterisk tech on a permanant basis though |
20:52.02 | NEEDINGHELP123 | anyone up for the job? |
20:52.11 | citywok | you might want to tell everybody where the job is located |
20:52.20 | NEEDINGHELP123 | remotely |
20:52.24 | NEEDINGHELP123 | whever you want it to be located |
20:52.26 | NEEDINGHELP123 | just be available |
20:52.32 | NEEDINGHELP123 | and ready to work on demand |
20:52.38 | NEEDINGHELP123 | with a minimum retainer fee |
20:52.40 | b11d` | im game :) |
20:52.41 | *** part/#asterisk LemensTS (~LemensTS@adsl-70-238-154-222.dsl.stlsmo.sbcglobal.net) |
20:52.47 | b11d` | haha no there are much better people for the job |
20:53.13 | citywok | i'm imagine there are at least a few people in here that would be happy to work out a gig with you. |
20:53.25 | b11d` | yeah, no shortage of Asterisk consultants, thats for sure :p |
20:53.25 | nny | <-- always willing to do remote support |
20:53.32 | citywok | if you have more details feel free to PM me :) |
20:54.22 | NEEDINGHELP123 | okay |
20:54.27 | NEEDINGHELP123 | so'll be in touch |
20:54.31 | NEEDINGHELP123 | I'm here and around |
20:54.44 | NEEDINGHELP123 | I'd like to get a few cv's with some info if possible |
20:54.49 | NEEDINGHELP123 | so that I can assertain who I need |
20:55.15 | b11d` | i'll be a phone sex worker, if need be |
20:55.48 | NEEDINGHELP123 | where is Drmessano |
20:55.53 | NEEDINGHELP123 | ? |
20:56.04 | NEEDINGHELP123 | he's a funny one where is here |
20:56.07 | NEEDINGHELP123 | not seen him for some time |
20:56.10 | NEEDINGHELP123 | :P |
20:56.18 | NEEDINGHELP123 | not been flmaed by him for some time that is :P |
20:56.26 | *** join/#asterisk Carp1 (~none@cpe-24-92-37-23.nycap.res.rr.com) |
20:58.59 | nny | looks like 5 pm, and for once I have no work load for the evening. guess I can leave the computer alone for a bit... later all |
20:59.20 | *** part/#asterisk nny (~Scott@cpe-071-076-058-253.sc.res.rr.com) |
21:03.06 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
21:13.37 | b11d` | ... |
21:16.37 | b11d` | just bought a new place... gotta go over to lowes, get a lawmmower.. weed eater... hedge trimmer.. |
21:17.03 | b11d` | why do all new mowers come with the insufferable bag on them.. i hate bagged mowers |
21:17.09 | *** join/#asterisk Benwa (~Benwa@unaffiliated/benwa) |
21:17.23 | Qwell | b11d`: you can use mine. you have to mow my lawn though. |
21:17.29 | b11d` | you got it |
21:17.58 | b11d` | i asked one of the neighborhood kids to mow the lawn... their mother overheard me saying "so how much do you charge?" -- then the cops got involved.. |
21:18.28 | wcselby | lol |
21:18.54 | wcselby | wait a week or two, you'll probably have a flyer hanging on your door with the name of a lawn service on it |
21:19.07 | b11d` | aye, you're likely right |
21:19.46 | ChannelZ | get a mulching mower.. mine has no bag |
21:19.54 | b11d` | yeah thats what im gonna get tonight |
21:20.30 | ChannelZ | is about to have 6 tons of landscape rock dumped in his driveway |
21:20.41 | b11d` | theres an 8-horse, 22" for $150 in town |
21:20.42 | ChannelZ | congrats on the new place btw |
21:20.45 | b11d` | thanks |
21:21.00 | ChannelZ | first house or something? |
21:21.08 | b11d` | yeah... |
21:21.12 | ChannelZ | cool |
21:21.19 | b11d` | got married, had a kid, needed a bigger place now |
21:21.24 | b11d` | working on kid #2 now |
21:21.40 | ChannelZ | still looking for parts? :) |
21:21.44 | b11d` | getting the 'hell years' done in short order :) |
21:22.03 | pigpen | does anyone know of a way to find out what telco has a porting agreement with? ie: I want to move my number.....and I want one friendly with asterisk. |
21:22.21 | citywok | i've ported to bandwidth.com and flowroute without any issue |
21:22.33 | ChannelZ | Isn't it law any of them have to port? |
21:22.53 | citywok | yea, there's no "agreement". the only requirement is the place you are porting the number to have a local presence in the area. |
21:22.54 | pigpen | they must port, but not all can port to anyone. |
21:23.19 | citywok | i was unable to port my san francisco numbers to qwest b/c they dont have a presence in the sanfran area |
21:23.22 | b11d` | yeah I recently ran into this exact problem |
21:23.38 | citywok | if they are TFN's, then it's really easy and doesnt really matter at all. |
21:23.45 | ChannelZ | I did Qwest -> Vitelity but they're both local. |
21:23.55 | ChannelZ | Qwest -> Integra too, same there though |
21:24.04 | citywok | ChannelZ: where are you? PNW? |
21:24.09 | ChannelZ | Colorado |
21:24.17 | citywok | ah. home of qwest. lol |
21:24.37 | ChannelZ | yeah. Soon to be Centurylink I guess. |
21:24.47 | citywok | yea. i got that email from our rep a few months ago. |
21:24.56 | b11d` | what? they are changing their name? |
21:24.59 | citywok | we run our entire company on qwest MPLS |
21:25.03 | ChannelZ | Centurytel is buying them |
21:25.03 | citywok | b11d`: they got bought |
21:25.06 | b11d` | ohhh |
21:25.13 | b11d` | im surprised |
21:25.30 | citywok | with any luck they will improve qwest support. |
21:25.33 | ChannelZ | they are a money toilet I think (qwest) |
21:25.40 | ChannelZ | so not sure why they bought them either |
21:25.43 | citywok | i'm praying that they rip out QControl and beat it like a red headed step child. |
21:26.08 | ChannelZ | but they have q.com! |
21:26.27 | b11d` | never even heard of Century Tel.. |
21:26.36 | b11d` | CenturyLink that is |
21:26.57 | citywok | i only have b/c my grandma's hosue had century-tel for phone service a few moons ago. |
21:27.34 | citywok | but i thought centurytel was smaller than qwest, i had no idea they were capable of buying them. |
21:28.10 | b11d` | well, I'm gonna go get that mower.. talk to you all tomorrow :0 |
21:28.11 | b11d` | goodnight |
21:28.39 | ChannelZ | have fun |
21:29.08 | ChannelZ | CenturyTel changed their name recently to CenturyLink. But I'd never heard of them either until I did a name-change video for them :) |
21:31.38 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
21:32.16 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
21:32.33 | boodu | hello |
21:32.59 | lvlolvlo | century.... *shrug* |
21:34.03 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
21:34.56 | IamTrying | What can i change in host=localhost? [Jul 29 23:30:57] NOTICE[15053]: chan_sip.c:20603 handle_request_register: Registration from '"xxx" <sip:xxxx@127.0.0.1>' failed for '127.0.0.1' - Peer is not supposed to register |
21:35.50 | [TK]D-Fender | IamTrying: You do no specify a host for a device that is expected to REGISTER |
21:36.33 | IamTrying | host=dynamic but i wanted to make a localhost test |
21:36.59 | [TK]D-Fender | IamTrying: Think of it like as if I was sure I knew your phone number, I would reject someone telling me where to contact you because I already know |
21:37.20 | [TK]D-Fender | IamTrying: Well your client is trying to register. So if you want it to succeed it MUST be "dynamic" |
21:37.34 | IamTrying | right, ok |
21:37.44 | IamTrying | [TK]D-Fender, thank you |
21:38.44 | *** join/#asterisk NuclearLucifer (gavroche@gavroche.pl) |
21:39.55 | *** join/#asterisk CoffeeIV (~rgr@dsl093-217-226.aus1.dsl.speakeasy.net) |
21:40.14 | *** join/#asterisk rayk_sland (~rklassen@mail.mccscs.com) |
21:41.21 | CoffeeIV | I have some Polycom phones that were registering to a asterisk box, and the asterisk box was moved to a new IP and the DNS changed accordingly, and now they don't register. I need some debugging tips -- is there a way I do a "udp telnet" to make sure port 5060 udp is open between the phones and the new server ? |
21:41.21 | Nugget | telnet is eeeeeeevil! |
21:41.46 | Katty | checks Nugget for signs of life |
21:41.49 | ChannelZ | sipsak? |
21:41.55 | CoffeeIV | evil problems require evil solutions |
21:41.58 | Chainsaw | CoffeeIV: You need to give up this telnet addiction. |
21:42.06 | *** join/#asterisk amacgyver (~macgyver@ns0.calibre-solutions.co.uk) |
21:42.07 | Chainsaw | CoffeeIV: Enable sip debugging for the phone IP address on the Asterisk console. |
21:42.18 | Chainsaw | CoffeeIV: Soft-reboot your phone. Sit back and watch. The server will tell you. |
21:42.21 | Katty | friends don't let friends telnet. |
21:42.22 | CoffeeIV | ok |
21:42.33 | Chainsaw | CoffeeIV: Add beer for additional relaxation. |
21:42.50 | CoffeeIV | telnet is awesome, we should have never replaced with these "browsers". netcat is also good. |
21:43.25 | *** join/#asterisk guilhermebr (~Guilherme@189.63.87.234) |
21:43.51 | ChannelZ | GOPHER! |
21:44.31 | NuclearLucifer | CoffeeIV, Have you got some dnscache in your network? Maybe it hasn't update yet? |
21:45.05 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
21:45.18 | Katty | hi raden |
21:46.15 | *** join/#asterisk guilhermebr (~Guilherme@189.63.87.234) |
21:46.21 | raden_work | hugs Katty |
21:46.39 | Katty | hugs on raden_work |
21:47.19 | *** join/#asterisk Mhaddog_Mac (~anonymous@adsl-32-170-204.mia.bellsouth.net) |
21:47.36 | CoffeeIV | I made sure the dns was good. Using "sip debug ip <mylocalip>" I can see the SIP register messages coming from my phone to the asterisk. |
21:48.50 | CoffeeIV | but "sip show registry" and the phone both indicate the registration is not happening. I don't see any log messages as to why. |
21:49.37 | *** join/#asterisk ltd_wk (~z@sixified.transact.net.au) |
21:49.58 | Chainsaw | CoffeeIV: You can observe the traffic now. I'm sure Fender would love a pastebin of the dialog. He's fluent in SIP. |
21:50.56 | raden_work | CoffeeIV, set sip debug on |
21:51.01 | raden_work | then have the phone register |
21:51.08 | raden_work | paste debug |
21:51.19 | CoffeeIV | ok, I'll do that and I'm setting up the pastebin |
21:54.58 | raden_work | Holly crap the dude actually paid me i helped the other day :) |
21:55.04 | raden_work | there are honest people in the world |
21:55.25 | NuclearLucifer | ha, I forced faxes to work with t38 today. And it works without any hangups. :) |
21:56.20 | raden_work | NuclearLucifer, really ? |
21:56.50 | CoffeeIV | here is my pastie: http://pastie.org/1066238 the IP of the phone that is trying to register is 10.0.2.9, the IP it is coming from is 66.93.217.226, the extension is 8449 |
21:57.07 | NuclearLucifer | raden_work, Yes, looks stable. But it would go `on production' in 1-2 weeks. But as far as I made tests, it works fine. |
21:57.23 | raden_work | I can never get it to work |
21:57.44 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
21:58.21 | raden_work | how can the ip of the phone and where it coming from be 2 different addresses ? |
21:58.57 | NuclearLucifer | raden_work, It's done with Asterisk 1.6.2.10. |
21:59.18 | raden_work | im on 1.6.0.10 |
21:59.23 | raden_work | ill have to try it out |
21:59.31 | raden_work | lemee check colorado co-los see what they running |
21:59.40 | NuclearLucifer | raden_work, I tested it from sending faxes from PSTN and to PSTN. |
21:59.52 | CoffeeIV | NAT, normally I would only use iax2 with NAT, but supposedly this was working previously. It's a really fancy HP super-router that can proably be configured to handle that kind of stuff |
22:00.08 | raden_work | 1.6.0.22 |
22:00.39 | raden_work | CoffeeIV, what are you trying to do exactly and what exactly is the issue ? |
22:02.00 | NuclearLucifer | raden_work, I connected fax to asterisk via linksys spa3021 voip gateway. |
22:02.06 | NuclearLucifer | And it works fine. |
22:02.23 | [TK]D-Fender | CoffeeIV: Smart routers are usually fucking DUMB. Just FYI |
22:02.36 | *** join/#asterisk Deathvalley122 (~Death@unaffiliated/deathvalley122) |
22:02.40 | CoffeeIV | I had an asterisk server on the 10.* internal network; a clone of it on an outside IP has been used as a backup. The internal one died, and I switched the DNS to point to the other IP, and everything seems to work -- including incoming calls to the new box -- except my phone doesn't register to the new external box, and thus can't make or receive calls. |
22:02.58 | [TK]D-Fender | CoffeeIV: <--- SIP read from UDP:66.93.217.226:38572 ---> |
22:03.06 | [TK]D-Fender | CoffeeIV: <--- Transmitting (no NAT) to 10.0.2.84:5060 ---> |
22:03.15 | [TK]D-Fender | CoffeeIV: And you blatantly did your configs wrong |
22:03.22 | CoffeeIV | Yeah, I hate smart routers too, and I would not have used SIP ever again if I had my way |
22:03.25 | [TK]D-Fender | CoffeeIV: its responding to the Private IP. |
22:03.41 | [TK]D-Fender | CoffeeIV: "no NAT" <- *cough* |
22:03.46 | [TK]D-Fender | CoffeeIV: Now go fix your peer |
22:07.58 | *** join/#asterisk seanjohn (~seanjohn@gateways.sheltoncomputers.com) |
22:08.26 | CoffeeIV | [TK-]D-Fender: setting nat=yes; in sip.conf did it, thanks so much. |
22:08.56 | CoffeeIV | you guys are awesome |
22:10.04 | mmlj4 | keep studying, soon you can be promoted to CoffeeV |
22:10.22 | raden_work | CoffeeIV, configure your router properly |
22:10.30 | *** join/#asterisk nettie (~nettie@stewie.freax.it) |
22:11.46 | nettie | Hi guys, anyone using Sangoma BRI cards here? I'm having issues with one of them. I think it's related to clock source I just hear a very distorted noise when I try to place/receive a call. |
22:11.52 | seanjohn | I have been pulling my hair out trying to fix something that is my termination provider's fault. If I use one of their other three servers, call quality is fine but using the server I had as the DEFAULT for incoming and outgoing, call quality crashed after 4 seconds of connectivity. I had complained to them before when they, voip.ms, had a problem (someone monitored several of my calls to toll free numbers) and they had the response "do |
22:11.52 | seanjohn | the premium route?" and "we are a respectable company". Respectable company my ass! A respectable company based on Finland (I'm in the US) would abide by the law and make sure their systems were secure no matter how much we paid for the call; this is illegal! |
22:17.42 | *** join/#asterisk seanjohn (~seanjohn@gateways.sheltoncomputers.com) |
22:18.00 | seanjohn | anyone else use voip.ms? |
22:18.14 | seanjohn | sorry, my workstation is REALLY failing (freezing) |
22:18.23 | seanjohn | windoze |
22:18.39 | seanjohn | unlike the asterisk machine |
22:19.36 | seanjohn | asterisk was causing hard drive errors to display when the specific server was malfunctioning |
22:20.04 | seanjohn | without asterisk running, the hard drive errors didn't display |
22:20.04 | Qwell | hard drive errors? VoIP provider? what? |
22:20.13 | seanjohn | yep Qwell |
22:20.15 | Qwell | secure? illegal? |
22:20.36 | seanjohn | I've been backing up everything for an hour and trying to fix anything on Centos for the past day |
22:20.49 | Qwell | how are any of these 4 things related? |
22:20.51 | citywok | seanjohn: why dont you describe your problem in a way that makes sense. so far your ranting makes none at all. |
22:24.36 | seanjohn | i'm trying to find the log lines |
22:24.43 | seanjohn | why doesn't asterisk rotate them more often |
22:24.51 | *** join/#asterisk Arsenick (~y@modemcable022.82-21-96.mc.videotron.ca) |
22:25.31 | seanjohn | I wish there was an admin here cause this is a bug in asterisk to asterisk sip (or callweaver) |
22:25.59 | seanjohn | my system was fine and my termination's "asterisk" was malfunctioning. |
22:26.06 | seanjohn | causing my system to malfunction to |
22:26.22 | seanjohn | the admin could tell me what I posted 3 entries into the room ago |
22:26.36 | seanjohn | fender??? |
22:26.37 | [TK]D-Fender | enables defragmentation on seanjohn |
22:26.43 | seanjohn | defrag? |
22:27.12 | seanjohn | sorry for pressing enter so much but I'm so pissed |
22:27.15 | [TK]D-Fender | seanjohn: 2 words summarize you right now : mixed nuts |
22:27.31 | [TK]D-Fender | seanjohn: Its coming out a garbled mess |
22:27.43 | seanjohn | fender, I came in here about 2 pm eastern and posted an exact error I was getting on asterisk |
22:28.16 | [TK]D-Fender | seanjohn: That is 4.5 hours ago. Long gone. |
22:28.56 | [TK]D-Fender | seanjohn: And the other side being callweaver... well fear not, there is already a special circle in Hell for that (Dante missed a few) |
22:29.16 | seanjohn | something about when the call hangs up on chan_sip.c "method not supported" and when this error appeared on asterisk, because of the remote server's malfunction, it would cause my system to produce fake write error messages. |
22:29.43 | seanjohn | i'm grepping all over the place |
22:30.04 | Qwell | fake write errors? |
22:30.31 | seanjohn | well, Qwell, e2kfts and badblocks said no errors, just fragmented |
22:32.42 | seanjohn | bingo |
22:32.43 | seanjohn | chan_sip.c: Remote host can't match request CANCEL to call '2245d2e93e4501a50e27c20d0b3bf4dd@173.50.101.11'. Giving up. |
22:33.39 | Qwell | and how does that have anything to do with disk write errors? |
22:33.49 | Qwell | or being secure, or...illegal? |
22:34.11 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
22:34.25 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
22:34.56 | seanjohn | it actually made the drive think it was having errors itself, storing a count of 9 errors in SMART, and producing the error ata2.00: cmd c8/00:08:8d:f7:b6/00:00:00:00:00/e5 tag 0 dma 4096 in |
22:34.56 | seanjohn | Jul 29 16:36:41 server1 kernel: ata2.00: cmd c8/00:08:8d:f7:b6/00:00:00:00:00/e5 tag 0 dma 4096 in |
22:34.56 | seanjohn | Jul 29 16:36:41 server1 kernel: res 51/40:08:8d:f7:b6/00:00:00:00:00/e5 Emask 0x9 (media error) |
22:35.00 | seanjohn | sorry |
22:35.24 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
22:35.25 | *** join/#asterisk italorossi (~italoross@201.76.154.130.intranet.digi.com.br) |
22:35.32 | citywok | are you saying a SIP message made your hard drive have a smart error? |
22:35.34 | Qwell | it IS having errors. |
22:35.48 | seanjohn | no, all other diagnostic tools say its fine |
22:35.59 | seanjohn | the 9 errors were produced between today and yesterday |
22:36.13 | Qwell | your diagnostic tools are wrong. |
22:36.27 | seanjohn | the same time as this started with their ONE, not the other three though, servers |
22:36.31 | citywok | smart errors dont always mean the drive is failing. |
22:36.36 | citywok | it could be remapping sectors |
22:36.56 | seanjohn | yes, citywok, but this is a raid 5 array |
22:36.57 | citywok | seanjohn: did you pass 3rd grade english? i'm having a hard time believing you did. |
22:37.00 | Qwell | seanjohn: log files write to disk. if you are logging errors, you are writing to disk. and those writes are causing *actual* errors. |
22:37.24 | seanjohn | Device: /dev/sda, 3 Currently unreadable (pending) sectors |
22:37.29 | citywok | seanjohn: i suggest you look up the difference between Causation and Correlation |
22:38.12 | seanjohn | i have had this happen before citywok with completely different hardware caused by a faulty LOCAL sip device. |
22:38.26 | seanjohn | its an exploit waiting to happen for dos of asterisk |
22:38.35 | citywok | what kind of hardware are you using? |
22:38.44 | seanjohn | i WAS using an spa2102 |
22:38.52 | citywok | pentium 133? software raid5? |
22:39.04 | seanjohn | that one went in the trash and no errors until now |
22:39.26 | seanjohn | citywok: q 8600 4 gb ddr2 |
22:39.37 | *** join/#asterisk Mhaddog_Mac (~anonymous@adsl-32-170-204.mia.bellsouth.net) |
22:39.38 | seanjohn | the 3 raid drives are western digital |
22:40.22 | seanjohn | the problem with the device that went in the trash was on an Athlon 3000 512 ddr |
22:40.34 | seanjohn | same error different hardware, completely! |
22:41.20 | seanjohn | this is asterisk 1.4.31 |
22:41.42 | seanjohn | x86_64 |
22:42.07 | citywok | well, i suggest you submit a ticket in the asterisk bug tracker if you believe this is a bug in asterisk. |
22:42.24 | Qwell | It's not a bug in Asterisk. |
22:42.29 | citywok | Before doing that i'd recommend you get a grasp of your actual problem, get all of the logs you can, compile with DONT_OPTIMIZE as htey suggest, and post the bug that way |
22:42.46 | citywok | Qwell: yea, i would put up a lot of money at very high odds that you are right :) |
22:42.47 | seanjohn | its not a bug, its caused by an idiot on the other end using a faulty build of asterisk and my asterisk doesn't counteract the issue |
22:42.57 | Qwell | citywok: No, I am definitely right. Period. |
22:43.07 | Qwell | Applications *CANNOT* cause write errors. |
22:43.11 | Qwell | They are caused by *ACTUAL ERRORS* |
22:43.17 | *** join/#asterisk l337ingDisorder (~l337ingDi@S0106000c425431b2.gv.shawcable.net) |
22:43.20 | citywok | yea. Causation and COrrelation as i suggested he look up. |
22:43.23 | Qwell | Your disks are broken. Fix them. |
22:43.30 | citywok | Correlation being the errors happen on writes. lol. |
22:43.40 | seanjohn | when I disabled this server and used the other 3, call quality is fine |
22:43.49 | Qwell | I wonder why. |
22:43.59 | seanjohn | not my server, one of their 4 servers |
22:44.07 | citywok | maybe during the time you tesetd against the other servers the disk errors were not being hit. |
22:44.19 | Qwell | how does call quality have anything at all to do with disk errors? |
22:44.23 | Qwell | you're making no sense. |
22:44.28 | seanjohn | tests were for 30 minutes, within 4 seconds of EACH call this would start |
22:44.31 | citywok | you should try writing a ton of garbage data to your disk and see if it throws smart errors |
22:44.43 | seanjohn | ok, command? |
22:44.50 | seanjohn | dd if= size= |
22:44.51 | seanjohn | ? |
22:45.00 | citywok | however you'd like, i dont give a shit |
22:45.10 | Qwell | Hire a consultant. |
22:45.24 | l337ingDisorder | I think I may have borked something. I'm following this guide: http://hubpages.com/hub/Installing-Asterisk-NOW-and-Configuring-Soft-Phones and it says after the AsteriskNow installer finishes, I should see a console screen where I can configure Asterisk. Instead I just get a login prompt and when I log in I get a standard bash prompt. I can access the web interface but it only has options... |
22:45.25 | citywok | i'd suggest hiring somebody that has a clue what they are doing. you clearly do not. |
22:45.26 | l337ingDisorder | ...for Voicemail & Recordings, Flash Operator Panel, and FreePBX Administration (According to the guide there should be a menu with a 'users' option). Can anyone help? |
22:45.30 | seanjohn | citywok, you act like I'm being arrogant to you or something, like you're the owner of voip.ms |
22:45.44 | citywok | i'm acting like you have no idea what you are talking about whatsoever |
22:45.48 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
22:45.49 | citywok | because you sound ignorant, not arrogant |
22:45.53 | seanjohn | go try it for yourself |
22:46.01 | seanjohn | put 5 bucks on voip.ms |
22:46.06 | seanjohn | use their chicago server |
22:46.32 | citywok | i'd bet if it were a problem with their server any of their thousands of other customers would have found it, and they would have resolved it within an hour. |
22:46.48 | citywok | so i'm going to save my time, and save you the hassle of paypalling me the $5. |
22:46.49 | seanjohn | its just the ONE server |
22:47.15 | seanjohn | you don't realize you can ORIGINATE to each server simultaneously but "i don't have a clue" |
22:48.53 | seanjohn | btw, the chicago and atlanta are the newest ones |
22:49.09 | seanjohn | different providers never used before |
22:49.10 | Qwell | So stop using the Chicago server |
22:49.20 | Qwell | problem solved. until your disk dies. |
22:49.25 | seanjohn | I did but I"m trying to help you in case this may be an explioit |
22:49.27 | citywok | lol, i really like that. |
22:49.33 | Qwell | an exploit? |
22:49.41 | Qwell | you're going to have to explain that |
22:49.58 | seanjohn | in case the way their service is malfunctioning, someone else doesn't repeat the same packets in an attempt to dos/ddos |
22:49.59 | citywok | qwell, he already did! the other user can send a SIP CANCEL that will cause your hard drive to fail. it's a DDoS. DUHHH!!!! JEEEZZZZ |
22:50.17 | Qwell | how is that going to DoS anything? |
22:50.37 | seanjohn | citywok, you couldn't have any sense other than computers if god touched you himself |
22:50.49 | TJNII | "SIP CANCEL that will cause your hard drive to fail." lol whut? I'm abviously late to the party, but.... |
22:50.49 | seanjohn | bash people, see where it gets you |
22:50.56 | Qwell | seanjohn: Is that what you're suggesting? |
22:51.09 | Qwell | is that a provider can cause a disk to fail by sending a bad SIP message? |
22:51.20 | seanjohn | i didn't post the messages online |
22:51.31 | seanjohn | from others having the same error and results |
22:51.41 | Qwell | TJNII: I've been here the whole time, and I still don't know wtf he's talking about. |
22:52.02 | Qwell | I'm pretty sure citywok nailed it though. |
22:52.05 | citywok | TJNII: neither do i. scroll up a bit if you have history. it's like a 3rd grader trying to explain what is going wrong. |
22:52.21 | *** join/#asterisk Arsenick (~y@fedora/Arsenick) |
22:52.34 | Qwell | seanjohn: Answer please. Is that what you're suggesting? That a provider can cause a disk to fail by sending a bad SIP message? |
22:52.36 | citywok | yea, i think i deduced what he was trying to get at. sip cancel causes logging that blows up the hard drive as a DDoS. it's a bug. maybe a cockroach. |
22:53.57 | [TK]D-Fender | http://tinyurl.com/37vuqsl <------------- |
22:54.02 | TJNII | ...interesting. |
22:54.13 | citywok | [TK]D-Fender: that's perfect. |
22:54.15 | Qwell | seanjohn: I'm trying to understand... |
22:54.39 | [TK]D-Fender | Qwell: Your FIRST mistake ;) |
22:54.40 | seanjohn | i guess you'll say that viruses never had access to CMOS rom |
22:55.08 | Qwell | seanjohn: IS that what you are suggesting? |
22:55.39 | joako | I setup chan_mobile ok call comes in and asterisk answers... but no audio. |
22:56.09 | Qwell | seanjohn: we can't fix it if we don't know what the problem is. |
22:56.26 | seanjohn | how secure do you think RTP is, especially when both the disk and dahdi, of asterisk, require the system clock to time theirselves. If asterisk threw off the timing somehow, it would cause disk errors |
22:56.37 | Qwell | oh good lord. I'm going home. |
22:56.47 | citywok | no kidding |
22:56.58 | citywok | soon we'll be hearing about conspiracy theories. did you know that 9/11 didn't happen? |
22:57.46 | TJNII | I have a coworker who is into conspiracy theories. He tried to tell me that the standardization of A 440 in music was due to space travel. I laughed in his face, couldn't help it. |
22:58.47 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
22:59.28 | TJNII | seanjohn: If you believe that, then reproduce it. Sniff the traffic that causes the failure, and create a reproducible testcase. |
22:59.44 | *** join/#asterisk edguy3 (~edguy@c-98-221-27-224.hsd1.nj.comcast.net) |
22:59.56 | citywok | make sure you read the asterisk bug tracker guidelines on posting a bug. |
23:00.23 | Qwell | seanjohn: issues.asterisk.org - please open an issue with a *CLEAR AND CONCISE* description. |
23:00.31 | Qwell | If it is not clear and concise, I will close it. |
23:00.55 | citywok | Qwell: if he does post it, can you PLEASE send me the bug#? it's going to be like comedy gold or something. |
23:01.48 | TJNII | If he can reproduce it, it is going to be a bug on many levels. |
23:01.55 | Qwell | seanjohn: I don't want to see any ranting about the provider, or Chicago, or fake errors. Give us detailed information, precise errors received, etc. No opinion. Mmk? |
23:01.58 | citywok | EBKAC? |
23:02.41 | Qwell | Nobody is going to take you seriously if you're just ranting and making no sense. |
23:03.19 | citywok | he still hasn't confirmed if my summary was what he was getting at |
23:03.31 | citywok | probably because when you make it 10 words long it sounds absurd |
23:03.40 | raden | walks in the room |
23:05.59 | citywok | walks in to a wall and bangs forehead chanting *it hurts to listen to seanjohn* over and over. |
23:07.47 | Qwell | seanjohn: Once you've reported the issue, /msg me the issue number so I can look at it. |
23:07.59 | TJNII | Eh, I'd just lay back and encourage him to reproduce it. He'll either give up, wise up, or prove us all wrong. |
23:09.16 | raden | Qwell, people with less intelligence than others usually do become an annoyance |
23:11.42 | *** join/#asterisk BlackBishop (dexter@d3xt3r01.tk) |
23:12.30 | BlackBishop | How can I give a busy signal if a user is already in a conversation ? ( like .. when I call myself ! ) |
23:12.34 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
23:13.12 | *** join/#asterisk chazzam (~chazz@173-24-236-101.client.mchsi.com) |
23:14.10 | joako | BlackBishop, you can set call-limit=1 |
23:15.01 | BlackBishop | I don't want all users not to be able to call anybody if one is calling .. |
23:15.24 | BlackBishop | I just don't understand why I don't get a busy signal if I call myself ! |
23:16.16 | beardy | Your phone has more lines than you currently use? |
23:17.19 | BlackBishop | I don't have an actual phone .. I'm just using SIP .. |
23:17.41 | beardy | phone/sip device/client/doodaa |
23:18.12 | BlackBishop | well, I guess not ! |
23:18.28 | BlackBishop | but that still doesn't explain why I don't get a busy signal when I call my extension .. |
23:18.56 | beardy | I think it does. |
23:19.22 | Kobaz | busy schmizzy |
23:19.37 | BlackBishop | nope |
23:19.38 | BlackBishop | on my mac |
23:19.41 | raden | when sip show peers is done does the port mean the port the device is registered on or the port its available at ? |
23:19.41 | BlackBishop | I can call myself ! |
23:19.46 | raden | 302/302 69.130.250.249 D N 26622 Unmonitored |
23:19.46 | raden | 303/303 66.168.15.100 D N 5060 Unmonitored |
23:19.46 | raden | 304/304 66.168.15.100 D N 1024 Unmonitored |
23:19.55 | BlackBishop | one sais ringing and the other says incomming call |
23:20.13 | BlackBishop | ( one window I meant ) |
23:20.20 | NuclearLucifer | BlackBishop, Set call-limit=1 under your sip context. |
23:20.23 | *** join/#asterisk Alagar (~Administr@122.164.32.124) |
23:20.27 | *** join/#asterisk philipp64|laptop (~chatzilla@75-92-150-245.war.clearwire-wmx.net) |
23:20.33 | NuclearLucifer | BlackBishop, As joako wrote. |
23:20.43 | beardy | Your sip client, whatever you want to call it, doesn't consider itself busy, and accepts another call (think two, or more, lines). |
23:20.49 | beardy | What they said. |
23:22.08 | BlackBishop | does call-limit limit outgoing calls ? |
23:22.15 | citywok | either you can make multiple calls, or yuo are busy. if you can make multiple calls, you aren't busy until you hit your call-limit |
23:22.18 | BlackBishop | what if in the future I'll try configuring "put on hold" .. |
23:23.22 | BlackBishop | there should be a way to only give the busy tone if I call myself ! |
23:23.47 | citywok | why would you need a busy tone for calling yourself? |
23:23.53 | citywok | that's the most retarded feature ever |
23:24.00 | beardy | You, and people should be clever enough not to do that. |
23:24.12 | BlackBishop | yeah, I did it just to see what happens .. |
23:24.35 | BlackBishop | and it doesn't happen like it does on my landline or cell phone |
23:25.07 | beardy | Becauese, like I said, you have more virtual incoming lines. |
23:25.09 | citywok | those don't allow more than one line to be in a "ringing" state |
23:25.12 | BlackBishop | citywok: I wouldn't consider it a feature .. but calling myself shouldn't be allowed either .. |
23:25.27 | citywok | a sip client doesnt mind if 5 lines are ringing at the same time |
23:26.12 | beardy | BlackBishop: Then you do some ugly hacking in your dialplan. |
23:26.35 | beardy | IF from own ext. KILL |
23:26.56 | citywok | yea, if CDR(SRC) = CDR(DST) playback(busy) |
23:27.11 | BlackBishop | goes read about dialplans |
23:27.31 | citywok | though i would consider that a non-issue and not waste my time with it |
23:29.40 | BlackBishop | well, it'll give me hours of "fun" to play with the dialplan ! :) |
23:29.54 | BlackBishop | 'till I get bored and go to sleep ( 2:30AM ) |
23:32.30 | beardy | What you want to do is just dumb it down to simulate a one-line POTS phone.. dumbing things down is almost always not very smart. |
23:33.04 | BlackBishop | how do I dumb it down if I only block calling myself ? |
23:33.31 | citywok | you are removing the ability for something to work |
23:33.37 | citywok | that's what i'd consider dumbing down |
23:35.01 | BlackBishop | I just don't see why I should be able to call myself ! and for the sake of fun at 2:30 AM because I can't sleep .. I'm just trying to see if I can do this .. |
23:35.04 | beardy | Configure your softphone to only use one "line" instead, if that's what you must have. |
23:35.41 | BlackBishop | the softphone doesn't have that "feature" ... |
23:36.12 | BlackBishop | wouldn't having "one line" also block putting calls on hold ? |
23:37.26 | beardy | Depends what you mean with that. Putting on hold and making more outgoing calls, no.. |
23:37.58 | beardy | But in general, lift your mind from POTS telephony limits. |
23:38.24 | BlackBishop | not only more outgoing, but more incomming .. |
23:39.06 | BlackBishop | wonders why irssi doesn't have a spellchecker by default |
23:39.44 | beardy | Because it would annoy 99% of its users, which would become 3 people if it did. |
23:39.59 | TJNII | Because some of us can spele properly, damnit! |
23:40.59 | BlackBishop | I didn't mean actually enabled by default .. but have the .perl script there .. |
23:41.05 | BlackBishop | sorry for the off-topic thingy :) |
23:46.13 | BlackBishop | exten => _X,1,GotoIf($["${CDR(src)}" = "${CDR(dst)}"]?Busy) |
23:46.28 | BlackBishop | now I gotta learn the syntax :) |
23:47.39 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
23:49.17 | *** join/#asterisk NuclearLucifer (gavroche@gavroche.pl) |
23:54.55 | BlackBishop | reload gets extensions.conf reloaded too, right ? |
23:55.27 | NuclearLucifer | right. |
23:55.43 | NuclearLucifer | But you can reload only extensions witt `dialplan reload'. |
23:57.07 | BlackBishop | noted, thanks :) |
23:57.36 | BlackBishop | now to figure out what exactly I'm doing wrong .. ( besides everything I'm trying to do ! ) |