| 00:00.29 | *** join/#asterisk lfod- (~email@unaffiliated/lfod-) |
| 00:05.27 | citywok | pabelanger: with the codec installed it does the same thing. no licenses = error. |
| 00:05.53 | citywok | pabelanger: that means if you run out of licenses you're in for trouble since it will not negotiate to ulaw. lol |
| 00:07.48 | jsjc | Ok seems that I goy FFA to work once! but i come into trouble. how can I have to force alaw or ulaw in the incoming calls? I have noload => codec_gsm.so but when it gets to the voicemail in a normal call asterisk goes bizarre! |
| 00:08.15 | citywok | jsjc disallow=all; allow=ulaw |
| 00:08.56 | jsjc | city seems that keeps gsm because cannot detect fax |
| 00:09.07 | jsjc | but If I do not load the codec detects fax straight away... |
| 00:09.13 | citywok | sip show channels, what codec does it say |
| 00:10.33 | jsjc | Format 0x0 |
| 00:11.18 | jsjc | sorry this one 0x80002 (gsm|h2 |
| 00:11.32 | citywok | look at the peer and see what codec it's configured for |
| 00:12.01 | jsjc | I have in general disallow=all allow=alaw,ulaw |
| 00:12.18 | citywok | yea but sip show peer |
| 00:12.22 | citywok | make sure it sees that |
| 00:13.04 | jsjc | 0x8000e (gsm|ulaw|alaw|h263) Codec Order : (alaw:20,ulaw:20) |
| 00:14.14 | jsjc | its confusing... why is not working... |
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| 00:48.54 | Gadfly | Hello all. I have a question about internal routing. I'm wondering if one asterisk installation listening on one dmz address could support multiple 'pbxes' (Multiple companies, reusing extensions)? I know I can handle the inbound routing based on DID, what I'm wondering is if I can have multiple extensions 201 (for example)? |
| 00:50.39 | Gadfly | To rephrase, is there analgous functionality to apache's name based virtual hosts? sip:201@companyA.com vs sip:201@companyB.com , where companyA.com and companyB.com resolve to the same IP |
| 00:58.35 | Gadfly | Is what I am referring to called 'SIP Domain suport' ? |
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| 01:15.07 | Toastie_ | Hello all. Posted to the list the other day but nothing as yet, thought I'd give this a shot. Anyone have suggestions as to how to prioritise calls across queues? |
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| 01:19.44 | Gadfly | Toastie_: basing the priority on what? |
| 01:21.02 | Toastie_ | Godfly: Basically extending the existing priority of a call within a queue, across multiple queues. Ie. A priority 100 call in Queue A should hit an agent before Priority 50 in Queue B |
| 01:21.31 | Toastie_ | But weights wont solve this as a Priority 100 in Queue B should hit an agent before a Priority 50 in Queue A (Ie. All queues are of equal importance) |
| 01:21.47 | ghenry | ho wto you kill a sip channels via cli? |
| 01:21.55 | ghenry | how do you, rather |
| 01:25.23 | Gadfly | Toastie_: why do you need separate queues than? |
| 01:25.41 | devmod | How can I get the codecs for the current call from the dialplan? |
| 01:26.03 | Toastie_ | Godfly: Skills based routing, except that some agents are skilled in multiple queue areas. |
| 01:26.10 | Gadfly | ah |
| 01:27.57 | Gadfly | so, how are you distributing calls? |
| 01:29.02 | Toastie_ | As in Queue stratergy? Ringall, and typically have to cascade two queues to deal with overflow seeing as there is no concept of a calling falling to a agent with a higher penalty based on timeout. |
| 01:29.14 | Toastie_ | *call |
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| 01:30.58 | ghenry | where does "sip history" store the SIP dialogs? |
| 01:31.10 | IsUp | ghenry: soft hangup <channel> |
| 01:31.21 | ghenry | yeah, forgot! thanks |
| 01:31.25 | ghenry | found it |
| 01:35.58 | Gadfly | well. I'm still pretty new to asterisk, but I think what you need to do is change the strategy to some logic so that the call with the highest priority (regardless of queue) that has a compatible agent available (skill based) is routed to that available agent. |
| 01:36.37 | IsUp | Gadfly: maybe you should use FastAGI and you can code yourself |
| 01:39.01 | Gadfly | IsUp: in reference to my question from earlier, or in reference to my suggestion? |
| 01:39.43 | IsUp | in reference to your suggestion |
| 01:42.33 | Gadfly | yes, FastAGI seems like the right choice |
| 01:42.55 | Toastie_ | Gadfly: Sorry, been spelling your name wrong, damn lack of autocomplete. To the best of my knowledge (Unless its changed recently) there is no strategy that will impact the call positions (For lack of a better term), and certainly not across queues. Overall though yes IsUp I think I'm going to have to whip something up as there doesnt seem to be anything to handle this nativly. |
| 01:44.33 | Gadfly | Toastie_: that's what I'm finding too |
| 01:45.03 | Gadfly | if your cdr data is in a DB, it should be pretty simple |
| 01:47.36 | Toastie_ | I'll have to do some testing, I know someone who is doing something similar but creating queues per call (Realtime) - it just sounds feral. |
| 01:51.10 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 01:57.31 | jsjc | exten => s,n,ReceiveFAX(${FAXFILE}.tif,fs) |
| 01:57.31 | jsjc | exten => s,n,System(/var/lib/asterisk/scripts/mailfax "${CALLERID(num)}" "${CALLEDFAX}" "${EXTNAME}" "${EXTEMAIL}" "${FAXFILE}" "${EXTCOMPANY}") |
| 01:57.40 | jsjc | when I do this seems like never gets to System |
| 01:57.51 | jsjc | could be that receiveFAX after success just finalises everything? |
| 01:58.13 | jsjc | and does not let go to next? |
| 01:58.16 | IsUp | jsjc: CLI output? |
| 01:58.32 | IsUp | core set verbose 10 |
| 01:58.44 | jsjc | isup lets check... |
| 02:06.57 | jsjc | == Spawn extension (macro-faxreceive, s, 11) exited non-zero on 'SIP/xxxxxxxxx-00000004' in macro 'faxreceive' |
| 02:07.13 | jsjc | without hitting the system one. |
| 02:09.56 | jsjc | oh perhaps i do not have that function in my asterisk.. |
| 02:09.57 | jsjc | lets see |
| 02:12.01 | pabelanger | ~pb |
| 02:12.01 | 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. |
| 02:12.05 | pabelanger | jsjc: ^' |
| 02:15.07 | jsjc | http://pastebin.com/VSHPhT0a |
| 02:15.21 | jsjc | sorry. Thought just last bit was needed. |
| 02:17.27 | *** join/#asterisk breardo (~chatzilla@234-200-29-134.hcc.mnscu.edu) |
| 02:17.49 | breardo | hey all.. anyone have a channel bank recommendation for ~200 analog lines? |
| 02:22.44 | breardo | looking at the ADIT 600 and Adtran CBs.. |
| 02:22.46 | breardo | anyone? |
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| 02:43.46 | IsUp | jsjc: http://www.voip-info.org/wiki/view/Digium+Free+Fax+For+Asterisk+and+sendEmail |
| 02:44.28 | IsUp | you should execute System on hangup |
| 02:44.33 | IsUp | h,1,System(... |
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| 03:28.15 | antiwire | is anyone who is awake in here using a Sangoma U100 "USBFXO" adapter? |
| 03:29.13 | WIMPy | ~ask |
| 03:29.13 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 03:29.24 | antiwire | oh my |
| 03:30.00 | antiwire | well my question was accurate. I really am asking if anyone in here is using one |
| 03:30.08 | antiwire | that's the real question |
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| 03:35.49 | Toastie_ | If I said I use one, would you ask another question? :) |
| 03:36.38 | antiwire | no, I'm just seeing if anyone really is using one, beside MemoServ |
| 03:36.39 | antiwire | me |
| 03:38.17 | Toastie_ | Well, there you go then lol. FWIW No, I don't actually use one. |
| 03:38.34 | De_Mon | anyone gotten odbc working with mysql 5.1? I'm trying to upgrade mysql and now func_odbc can't talk to it :( |
| 03:39.06 | antiwire | careful De_Mon, anyone is really vague, apparently |
| 03:39.29 | De_Mon | it's not vague it's broad |
| 03:40.55 | Toastie_ | antiwire I think you'll find that 1) Theres no need to get your knickers in a knot - no one was having a go at you heh 2) The problem with your question wasn't the anyone. It was the assumption you had another question.. |
| 03:41.14 | antiwire | oh man I'm not knicker knotted |
| 03:41.46 | Toastie_ | And here I was thinking things were about to get kinky. |
| 03:42.09 | IsUp | watching jerry springer show |
| 03:42.32 | antiwire | but I am worried since I still don't think anyone else on this planet is using a U100. by some anomalous fluke of nature I am the only one |
| 03:42.37 | antiwire | and it's loney |
| 03:42.52 | antiwire | ly |
| 03:45.27 | De_Mon | antiwire I'm not |
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| 03:45.59 | IsUp | hey WIMPy |
| 03:46.02 | De_Mon | mysql why do you hate me so |
| 03:46.16 | De_Mon | is it because I hated you first? I woln't say I'm sorry... |
| 03:46.41 | IsUp | whats wrong De_Mon |
| 03:46.42 | WIMPy | De_Mon: It's not only hardware. Software is evil as well. |
| 03:46.47 | IsUp | any output/error? |
| 03:46.48 | WIMPy | Hi IsUp |
| 03:47.18 | De_Mon | IsUp mysql 5.1 and odbc aren't playing nice |
| 03:48.22 | De_Mon | and either i'm not the only one that's having this problem or nobody wants to admit they use it |
| 03:48.58 | IsUp | what happens when u execute a query? |
| 03:49.28 | De_Mon | i'm just trying to get isql to connect and I get access denied. mysql client works just fine from the same box though |
| 03:51.05 | De_Mon | can you connect from odbc to mysql 5.1? |
| 03:51.26 | IsUp | i dont have both |
| 03:51.46 | IsUp | check your mysql logs |
| 03:52.21 | IsUp | tail -f error.log |
| 03:53.10 | De_Mon | mysql is on windows and doesn't appear to have any logging |
| 03:54.06 | De_Mon | i was going to restart it later tonight with logging enabled (I dont think it can be turned on while running), but considering mysql client works it seems like I shouldn't be the only one with this problem heh |
| 03:54.52 | IsUp | did you enabled '%' host on your mysql user? |
| 03:55.03 | De_Mon | yea |
| 03:55.13 | De_Mon | mysql client didn't work till I did that |
| 03:55.52 | IsUp | probably something wrong with your mysql setup |
| 03:56.15 | De_Mon | you don't think it's an odbc issue? |
| 03:56.43 | IsUp | no, as my opinion |
| 03:57.37 | De_Mon | uugh |
| 03:57.47 | De_Mon | maybe I can do this while mysql is running after all |
| 03:58.40 | IsUp | your pbx and db on LAN? |
| 04:01.47 | De_Mon | yeah |
| 04:03.25 | Toastie_ | If you can connect with mysqlclient but not ODBC - both on the same host - I'd be inclined to say ODBC (Or your connection string) is broken. |
| 04:05.38 | De_Mon | thats my thinking too Toastie_ |
| 04:06.20 | *** part/#asterisk breardo (~chatzilla@234-200-29-134.hcc.mnscu.edu) |
| 04:07.53 | De_Mon | well, i tried turning on logging in mysql but none of it is logging any connection attempts successful or failed :( |
| 04:09.32 | Toastie_ | If you fire up a tcpdump on your Linux box while trying to make an ODBC connection do you see packets going in both directions? |
| 04:10.28 | Toastie_ | (Its been a long time since I dealt with Unix ODBC but its possible its as stupid as it was way back and is just giving you a 403 for anything it cant handle) |
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| 04:19.38 | De_Mon | hrmph my libpcap and ngrep are broken |
| 04:20.03 | De_Mon | I did a bunch of updates while updating myodbc and evidently broke this one sec |
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| 04:23.01 | De_Mon | the error from odbc is directly from the server |
| 04:23.25 | De_Mon | i think the driver is sending the wrong authentication arguments or something stupid |
| 04:24.46 | De_Mon | i'm going to try the libs directly from mysql |
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| 05:01.53 | De_Mon | no dice time to fix connections to 4.0 and call it a night :( |
| 05:07.49 | nightwalk | Hmm...is there any easy way to get asterisk to show me why it's exiting prematurely when it tries to start up? I tried 'asterisk -d -T -vvv', and even tried strace'ing it to no avail |
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| 05:34.58 | pabelanger | nightwalk: asterisk -vvvcg |
| 05:35.08 | pabelanger | -g is to generate a core dump |
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| 06:52.16 | schmidts | good morning |
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| 07:05.09 | nightwalk | pabelanger: it isn't coring, that's kind of the problem. It's voluntarily exiting for some reason, and it isn't nice enough to tell me *why* even when I run it in debug mode :/ |
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| 08:41.38 | joachim_- | any1 familiar with heartbeat and Asterisk? |
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| 09:16.28 | maxagaz_ | Hi |
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| 09:39.19 | jmkgreen | Hi - anyone heard of problems with sipgate UK at all? Our * server lost contact at 4am gmt on the 19th. A tcpdump shows we're sending registrations and getting nothing back. |
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| 09:49.57 | tuxx- | jmkgreen: maybe contact sipgate? :P |
| 09:50.16 | jmkgreen | tuxx-: Easier said than done :-( |
| 09:50.28 | tuxx- | get a better sipprovider then |
| 09:50.28 | tuxx- | ;-) |
| 09:50.53 | jmkgreen | I'm assuming they've not gone under |
| 09:51.12 | jmkgreen | apparently we only use them for direct dial inbounds to people's desks so they aren't critical |
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| 09:51.49 | jmkgreen | our main numbers are isdn so that's not a problem. I just find it hard to believe they've just disappeared off the face of the earth (apparently) |
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| 11:10.44 | garymc | Hi Guys, anyone know anything about Dialers? |
| 11:12.18 | Cain | Hi, i want to specify rx/tx gain for every channel, so i modify chan_dahdi like this ==> http://pastebin.com/FB3SFfbc is there any mistake ? (TDM410P & Asterisk 1.6.2.15) |
| 11:12.20 | garymc | I've got a website that people put there phone numbers into. I then want a dialer to call them, then when they pick up it puts them through to my call centre. Anyone know how to do this? |
| 11:19.08 | Tozz_ | use .call files |
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| 11:30.27 | oelewapperke | my asterisk is opening up thousands of udp sockets for apparently no reason at all |
| 11:30.33 | oelewapperke | any ideas what this could be ? |
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| 11:44.32 | E-bola | is loving how asterisk 1.8 finally apears to be stable |
| 11:50.28 | garymc | Tozz : Are call files simple? I currently use freepbx can i integrate it? |
| 11:53.02 | Tozz_ | their easy. its a small text file you place in a directory |
| 11:53.09 | Tozz_ | asterisk checks for call files and executes them |
| 11:53.18 | Tozz_ | the call file contains data as to which number to dial, etc, etc |
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| 11:53.27 | Tozz_ | google for it and you'll see ;) |
| 11:53.38 | jpvoip | Hello, i want to customize Asterisknow (asteriskgui) creating a new page to the GUI.... where i can find a tutorial talking about this? |
| 11:59.49 | tuxx- | jpvoip: maybe ask in #asterisknow |
| 11:59.58 | tuxx- | omg, its snowing \o/ |
| 12:23.33 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 12:25.12 | jpvoip | tuxx-: i did, no answer yet :( |
| 12:26.59 | *** join/#asterisk fenrus (fenrus@drickheroin.se) |
| 12:26.59 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 12:27.05 | fenrus | i get, The device state of this queue member, SIP/403, is still 'Not in Use' when it probably should not be! |
| 12:27.09 | fenrus | Please check UPGRADE.txt for correct configuration settings. |
| 12:27.14 | wonderworld | hey, is someone experienced with misdn in here?? |
| 12:27.16 | fenrus | cant seem to find the UPGRADE.txt file :D |
| 12:31.01 | wonderworld | i have got a problem with an asterisk running with misdn on a PMP line. suddenly no incoming calls are possible anymore. http://pastebin.com/vu0vASLA |
| 12:31.26 | wonderworld | callers just hear the recording from the telco "this line can't be reached right now" |
| 12:31.36 | wonderworld | this happens every 2 or 3 month |
| 12:31.48 | wonderworld | any ideas? |
| 12:40.45 | fenrus | is that how it looks when its not working ? |
| 12:40.50 | fenrus | could be some clock issue |
| 12:41.54 | fenrus | im not sure how this is configured or called in theese cards, but i think that you should use the clock from the line since the telco equipment has better clocks than pc |
| 12:41.57 | fenrus | s |
| 12:43.13 | wonderworld | yes, thats the actual state |
| 12:44.15 | wonderworld | i din't set up this asterisk and i never used misdn before. so you think it might be an issue with "1,master_clock" ? |
| 12:48.39 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 12:49.16 | *** join/#asterisk ugenk (~ugenk@gate.mgts.by) |
| 12:49.53 | ugenk | hello! why when in queue() called hangs up, dialplan does not continue?.. even to h extension |
| 12:49.58 | joachim_- | any1 here using heartbeat with asterisk? |
| 12:50.34 | *** join/#asterisk mpe (~mpe@office.ipvision.dk) |
| 12:51.58 | joachim_- | I have it up and running OK, just wandering if this is accepted in a production system? Is it stable enough for asterisk, and what will happen to the sip registrations upon server switch? |
| 12:52.27 | fenrus | wonderworld, im not that good at e1 |
| 12:52.30 | fenrus | s and shit |
| 12:52.43 | fenrus | but that is my guess, ask the telco how they think it looks, if they can see any errors |
| 12:52.50 | tuxx- | joachim_-: your sip registrations are in astdb, you could let the heartbeat script copy that to your second machine, so the sip registrration stay registered |
| 12:52.54 | tuxx- | ;p |
| 12:53.27 | ugenk | or use realtime |
| 12:54.06 | joachim_- | tuxx-: Of course! Thanks for the tip. Using realtime all the way of course. Got 3 asterisk servers & 3 sql servers. All identical |
| 12:54.43 | ugenk | u need a kind of replication between mysql's |
| 12:55.03 | joachim_- | ugenk: yeah, I was thinking about csync2 |
| 12:56.15 | ugenk | ye |
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| 13:15.20 | joobie | anyone in the states / uk know of a stable / cheap provider to go with for ISDN ? |
| 13:16.37 | tzanger | you won't find that in the US |
| 13:16.59 | tzanger | north america kind of gutted its ISDN BRI service |
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| 13:53.16 | joobie | oh |
| 13:53.25 | joobie | what do you guys use then for digital voice? |
| 13:53.34 | joobie | what's the standard for like a 30 line business |
| 13:53.43 | *** join/#asterisk fc_ (~fc@94-193-100-127.zone7.bethere.co.uk) |
| 13:54.01 | fc_ | hi all |
| 13:54.16 | fc_ | i have a question about AGI |
| 13:54.33 | fc_ | is it possible to run an AGI script manually from asterisk CLI on 1.6.1 ? |
| 13:54.40 | fc_ | tried with >agi exec |
| 13:54.49 | fc_ | but not really understanding how to specify application and arguments |
| 13:54.53 | fc_ | any help appreciated :) |
| 14:00.59 | fc_ | anyone can help with AGI execution from asterisk cli? |
| 14:02.08 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:02.08 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:02.32 | mzahariev | as far as i know you can not do that |
| 14:02.50 | mzahariev | try setting up an extension to call the AGI |
| 14:03.00 | mzahariev | and from cli set |
| 14:03.01 | fc_ | thanks mzahariev :) |
| 14:03.03 | mzahariev | agi debug |
| 14:03.10 | fc_ | i actually already have one extension |
| 14:03.22 | fc_ | is a freepbx agi script for logout and login of user on the device |
| 14:03.29 | fc_ | i tried with ORIGINATE |
| 14:03.36 | fc_ | to create a call from my device to the extension |
| 14:03.39 | mzahariev | to see what is happening |
| 14:03.52 | fc_ | it works ... except that i have to pickup the call on the phone and so i can't really automate it |
| 14:03.58 | fc_ | i ll try |
| 14:04.15 | mzahariev | try Music on hold :) |
| 14:04.38 | fc_ | mzahariev: what do you mean? :) |
| 14:04.45 | *** join/#asterisk luckman212 (~quassel@pool-173-77-253-145.nycmny.fios.verizon.net) |
| 14:04.46 | mzahariev | or Answer() and then Wait() |
| 14:04.51 | fc_ | ah right |
| 14:04.54 | fc_ | i will :) thanks |
| 14:08.31 | wonderworld | fenrus: tnx |
| 14:09.42 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
| 14:16.39 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 14:19.09 | *** join/#asterisk gego (~quassel@b238085.customer.hansenet.de) |
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| 14:29.43 | *** join/#asterisk cherva (~cherva@78.128.100.50) |
| 14:31.40 | cherva | can someone help me I want to make my SIP tunks to have only one incoming line .... I tested by setting maximum channels to 1 and modifing the context to from-tunk-sip-NAME_OF_THE_SIP and I added call-limit=1 into peer details but nothing helped........ :? |
| 14:36.30 | E-bola | leifmadsen: I was wondering if all those relationships for 18511 was correct :) |
| 14:36.33 | *** part/#asterisk jpvoip (~c8af3dfa@redenorte.net) |
| 14:36.52 | leifmadsen | E-bola: a relationship doesn't necessarily mean much |
| 14:37.04 | leifmadsen | it's just, "hey, may be somewhat related" |
| 14:37.11 | leifmadsen | don't take them too literally |
| 14:37.24 | E-bola | ok |
| 14:37.34 | leifmadsen | cherva: use GROUP() and GROUP_COUNT() in the dialplan |
| 14:39.40 | titter | chan_sip.c:3633 __sip_xmit: sip_xmit of 0x2aaad40064f0 (len 538) to 72.86.xx.xx:0 returned -1: Invalid argument -- Been getting them flooding my CLI. Can't find much on it, anyone have any ideas? |
| 14:40.41 | IsUp | titter, you edited anything recently? |
| 14:40.43 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 14:40.47 | IsUp | in sip.conf? |
| 14:41.27 | fenrus | wonderworld, did you get any help from someone? :) |
| 14:43.45 | titter | Nothing out of the ordinary. the sip.conf is pretty slim, use includes to include other sip account files |
| 14:45.32 | IsUp | i had same problem yesterday, i was trying to include a file which not exists |
| 14:45.36 | *** join/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-226-170.w86-204.abo.wanadoo.fr) |
| 14:45.37 | IsUp | so restart helps alot :p |
| 14:45.58 | leifmadsen | yes, chan_sip won't load if you try to include a file that doesn't exist |
| 14:46.05 | titter | They exist. There is around 30 phones at the remote location. Rebooting their router fixed it. |
| 14:46.22 | titter | However it comes back |
| 14:46.29 | merlin8282 | Hi all. I'd need urgently an answer to this question : are there any differences between the asterisk 1.4 and 1.8 (mainly dialplan) configuration/syntax ? |
| 14:47.26 | IsUp | merlin8282: read CHANGES/UPGRADE files |
| 14:48.59 | *** join/#asterisk iulhk (~iulhk@175.110.28.210) |
| 14:49.24 | merlin8282 | IsUp: that's a huge amount of text :/ |
| 14:49.34 | IsUp | so? |
| 14:50.12 | Tozz_ | well doh! its friday! |
| 14:50.18 | Tozz_ | cant u read it for him?! |
| 14:50.30 | titter | Pfft text to speech |
| 14:50.30 | IsUp | i prefer Bible |
| 14:50.49 | merlin8282 | If I didn't want a quick answer I wouln't have asked here ;) It'd be nice if I simply have an answer like 'yes, there are syntax differences' or 'no, same syntax'. |
| 14:50.58 | IsUp | yes, there are syntax differences |
| 14:50.59 | merlin8282 | Of course I'll read it later. |
| 14:51.00 | titter | merlin8282: yes |
| 14:51.05 | merlin8282 | Ok, thanks. |
| 14:51.10 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 14:51.21 | IsUp | so if your boss is forcing to you to upgrade |
| 14:51.25 | IsUp | dont upgrade yet |
| 14:51.29 | Tozz_ | merlin8282: I tried upgrading from 1.4 to 1.6 |
| 14:51.31 | Tozz_ | wasn't that easy |
| 14:51.35 | Tozz_ | I reverted back to 1.4 |
| 14:51.38 | merlin8282 | right, that's the point IsUp |
| 14:51.49 | IsUp | yeah i can feel that pressure |
| 14:54.34 | andylockran | hey guys, just looking at writing a perl-ami app... however, am not able to connect using the Asterisk::AMI example. |
| 14:54.59 | andylockran | though telnet connections using same credentials work fine |
| 14:54.59 | Nugget | telnet is eeeeeeevil! |
| 14:55.19 | IsUp | always same comment from Nugget :p |
| 14:56.13 | IsUp | bot or something? hello my name is telnet i from telnet |
| 14:56.17 | IsUp | i love telnet |
| 14:56.21 | IsUp | umm.. |
| 14:56.33 | andylockran | haha, maybe not |
| 14:56.51 | andylockran | Nugget: only using it for testing.. never live. |
| 14:57.10 | andylockran | though have got some SCO servers someplace which require telnet... though I stay away from that. |
| 14:59.19 | *** join/#asterisk vader-- (vader@c-71-225-201-226.hsd1.nj.comcast.net) |
| 14:59.21 | vader-- | hmmm |
| 14:59.26 | vader-- | hola everyone |
| 15:00.16 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:00.16 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:00.20 | IsUp | hello vader-- |
| 15:00.31 | vader-- | have you guys ever seen where a cisco phone will hang up calls like after picking up the receiver? |
| 15:00.39 | vader-- | i have a user who is claiming their phone is doing that |
| 15:00.46 | vader-- | but when they goto show people it doesn't happen |
| 15:01.36 | andylockran | vader--: pebcak I think... |
| 15:01.45 | vader-- | pebcak? |
| 15:01.58 | Tozz_ | google that ;) |
| 15:02.15 | vader-- | haha ya |
| 15:02.18 | vader-- | ive heard that before |
| 15:03.04 | andylockran | the key 'goto show people and it doesn't happen' seems to be the most important factor there |
| 15:03.09 | WIMPy | Maybe a bouncing hook switch? |
| 15:05.42 | *** join/#asterisk dajhorn (~dajhorn@adsl-75-17-124-41.dsl.rcsntx.sbcglobal.net) |
| 15:06.05 | andylockran | anyone using Asterisk::AMI perl module? |
| 15:08.00 | *** join/#asterisk v1s (~v1s@202.84.107.67) |
| 15:08.04 | *** join/#asterisk vader-- (vader@c-71-225-201-226.hsd1.nj.comcast.net) |
| 15:08.45 | v1s | is this my server or some one trying to hack my server? I am getting a ton of these in the cli |
| 15:08.46 | v1s | [Jan 21 22:45:24] NOTICE[11236]: chan_sip.c:21639 handle_request_register: Registration from '"admin" <sip:admin@myservIP>' failed for 'UNKNOWN-IP' - No matching peer found |
| 15:09.11 | vader-- | ya i was thinking maybe the switch in the phone is beat |
| 15:09.17 | IsUp | someone is doing stress test on your server :p |
| 15:09.34 | v1s | trying to find a login? |
| 15:09.42 | andylockran | yeah, I've got a whitelist of allowed ips for my asterisk server. |
| 15:10.00 | andylockran | and fail2ban for IPS on that range if they fail authentication. |
| 15:10.35 | andylockran | only time I remove the whitelist and allow all (still with fail2ban) is when we've got workers travelling abroad.. |
| 15:10.36 | IsUp | v1s: yeah maybe, enable firewall right now. and check your logs |
| 15:10.46 | v1s | yes i blocked the ip already |
| 15:11.43 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-ytaguwckowujnipv) |
| 15:13.42 | andylockran | v1s: I really recommend fail2ban and it's asterisk jail. |
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| 15:18.12 | leifmadsen | its, not it's |
| 15:19.14 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:20.06 | WIMPy | Why do professional routers have to be so shitty? |
| 15:20.11 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:20.20 | WIMPy | was just synflooded offline several times. |
| 15:26.14 | *** join/#asterisk kaii (~kaii@p5DDA5AC7.dip.t-dialin.net) |
| 15:27.19 | *** join/#asterisk kaii (~kaii@p5DDA5AC7.dip.t-dialin.net) |
| 15:27.53 | kaii | i'm writing a script to convert dialplans from 1.4 syntax to 1.6/1.8 syntax, replacing all the ExecIf() calls and stuff |
| 15:28.21 | kaii | theres one point i'm unclear about .. regarding macros. (i replace all occurances of macros and macro calls with gosubs) |
| 15:29.10 | kaii | are all variables defined inside a macro considered local? so when the macro terminates, they dont exist anymore? if that is true, i should replace all Set(XY=) statements with Set(LOCAL(XY)=) inside the gosub |
| 15:29.55 | andylockran | thanks leifmadsen |
| 15:30.47 | WIMPy | kaii: Don't replace the macros. |
| 15:30.48 | leifmadsen | kaii: basically ya -- although you don't really need to convert your Macro()'s |
| 15:31.00 | leifmadsen | Macro() wasn't removed (and isn't being removed) |
| 15:31.09 | leifmadsen | ~gosub |
| 15:31.14 | leifmadsen | ~macro |
| 15:31.23 | leifmadsen | I should write something about this |
| 15:31.59 | kaii | maybe you really should |
| 15:32.18 | leifmadsen | other work to do first that pays :) |
| 15:32.19 | kaii | but wouldnt it be better to only use gosub instead of macro? |
| 15:32.26 | leifmadsen | define "better" |
| 15:32.30 | leifmadsen | why is it better? |
| 15:32.40 | leifmadsen | they are just different |
| 15:32.53 | leifmadsen | I use both in different situations |
| 15:33.58 | leifmadsen | you can convert them if you want to, but you don't have to |
| 15:34.06 | kaii | macro has this memory stack limitation .. which gosub doesnt have. what can you do with a macro what you cant to with gosub? |
| 15:34.16 | kaii | the UPDATE document clearly states: Macro() is now deprecated. If you need subroutines, you should use the Gosub()/Return() applications. |
| 15:34.35 | kaii | ok i understand i dont have to |
| 15:34.35 | leifmadsen | deprecated just means, "not updated" |
| 15:34.40 | kaii | i know |
| 15:34.48 | leifmadsen | anyways, not all variables are local in a macro |
| 15:34.51 | leifmadsen | just the ARGx ones |
| 15:34.51 | kaii | it also means "gosub is the future" |
| 15:35.02 | leifmadsen | which is the same in GoSub() |
| 15:35.19 | leifmadsen | oh wait, you said Set() in the Macro() |
| 15:35.23 | kaii | okay so if i set XY=foo inside of a macro, and return to the point where it was called, i still can access "XY" and get "foo"? |
| 15:35.28 | leifmadsen | then ya, you probably want to wrap those in LOCAL() |
| 15:35.37 | leifmadsen | heck, just test it :) |
| 15:35.56 | kaii | have no setup here at home office :/ |
| 15:36.14 | leifmadsen | VMware Player |
| 15:36.37 | WIMPy | A Goto with parameters would benice. |
| 15:36.47 | WIMPy | s/ben/be n/ |
| 15:36.48 | leifmadsen | that's what GoSub() is... |
| 15:36.49 | kaii | of course there are some special cases which i cant (or dont want) to convert automaticall .. such as macroexclusive() and calls to MACRO_CONTEXT etc. which i believe there is no equivalent in GoSubs .. i'll just warn on them |
| 15:37.16 | WIMPy | I don't intend to ever return. |
| 15:37.26 | kaii | then dont? |
| 15:37.28 | leifmadsen | so don't call Return() |
| 15:38.03 | leifmadsen | You can GoSub() several levels deep and it'll only return as many layers as you call Return() |
| 15:38.06 | WIMPy | I'm old. I don't like wasting resources on a return address I don't need. |
| 15:38.58 | leifmadsen | shrugs |
| 15:39.04 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
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| 15:42.18 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:42.55 | *** join/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 15:43.33 | nextime | hello all |
| 15:44.25 | nextime | asterisk 1.6, dynamic features. I have an appliaction in application map inside features.conf like this: |
| 15:44.26 | nextime | domotikaext1 => *1,self/callee,System(/usr/bin/domotikacmd ${DOMOTIKA_CMD1}) |
| 15:45.08 | nextime | my issue is tht the variable DOMOTIKA_CMD1 isn't passed to the system call even if i set on my dialplan before the dial |
| 15:45.29 | nextime | any suggestion? |
| 15:48.30 | kaii | i guess this is just because features.conf is not extensions.conf and you can not use dialplan variables in the system call |
| 15:50.13 | nextime | kaii: in the documentation inside features.conf it say that i can use channel variables, and core show channel SIP/xxx say that my variables are there |
| 15:50.44 | ketema | I am trying to get voicemail working with odbc and I am getting this error : /usr/lib/asterisk/modules/app_voicemail.so: undefined symbol: ast_odbc_direct_execute i think i might not have set my dsn up correctly, does anyone have a console app that I can test my odbc connection with ? |
| 15:51.30 | nextime | ketema : it seem more a compilation/environment library error to me |
| 15:52.26 | nextime | so, you have the wrong verion of odbc libraryes, not the same where the app_voicemail was compiled against |
| 15:52.51 | nextime | ( and something is changed in the api ) |
| 15:52.56 | *** join/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
| 15:53.35 | ketema | nextime: i thought so too, but i got no errors when compiling and I googled and found https://issues.asterisk.org/view_all_bug_page.php and from what i can understand if you have a syntax error in SQL somewhere or perhaps bad config it crashes...so i am trying to eliminate my possible errors first |
| 15:54.09 | ketema | nextime: if thats true how do I find out which version of ODBC i need ? |
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| 15:57.28 | andylockran | ok, so I'm getting rejected from the API in a consistent fashion, as per http://www.voip-info.org/wiki/view/Asterisk+manager+Example%3A+Originate would suggest. |
| 15:57.43 | andylockran | any ideas how to fix the problem? |
| 15:58.15 | *** part/#asterisk NotHere (~steve@geozix.static.otenet.gr) |
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| 16:05.14 | shamelessn00b | Hi guys, I'm configuring inward dialing on my * box, here's what I'd like to do, once someone dials from PSTN to my number, I'd like to retain the Caller ID of that user and relay it to a particular SIP user registered on my asterisk |
| 16:06.04 | WIMPy | shamelessn00b: That's what happens unless you actively alter the caller ID. |
| 16:06.38 | shamelessn00b | WIMPy: so its automatically relayed? |
| 16:08.18 | jkroon | no, it's automatically stored in ${CALLERID(num)} |
| 16:09.08 | shamelessn00b | same thing |
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| 16:26.33 | maxagaz__ | hi |
| 16:28.54 | maxagaz__ | someone free to have a little talk on TLS ? |
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| 16:31.25 | Katty | so i'm setting up ssh keys for two boxes. generated a key on box A, under root, now i want to copy it to Box B, so that they can copy some stuff back and forth when need be. I /think/ i just need to do ssh-copy-id root@boxbipaddress. is that right? or is there more to it? |
| 16:32.54 | ChannelZ | well you need the fingerprint on the other box |
| 16:33.14 | ChannelZ | in the authorized_keys for whatever user you're logging in as |
| 16:33.36 | IsUp | ssh-copy-id -i ~/.ssh/id_rsa.pub root@remotehost |
| 16:33.47 | ChannelZ | the public key half of the key you generated |
| 16:34.08 | Katty | well i did the ssh-copy-id root@boxbipaddress |
| 16:34.12 | Katty | and it looks ok |
| 16:34.27 | IsUp | yeap |
| 16:34.32 | Katty | i did an ssh root@foo and it didn't prompt me for a password |
| 16:34.39 | ChannelZ | hurray! |
| 16:34.45 | Katty | where do i go to see if the key is actually there tho |
| 16:34.48 | Katty | the txt file or whatever |
| 16:34.55 | ChannelZ | ~/.ssh/authorized_keys |
| 16:34.56 | IsUp | .ssh/authorized_keys |
| 16:35.06 | Katty | i don't understand those paths |
| 16:35.16 | IsUp | /root/.ssh/authorized_keys |
| 16:35.16 | ChannelZ | ~ = home directory of whatever user (root?) |
| 16:35.21 | Katty | oh, ok |
| 16:35.58 | Katty | i nanoed authorized_keys and it shows a big long list of stuffs |
| 16:36.04 | Katty | i'm guessing that's correct |
| 16:36.10 | ChannelZ | it'd be the last line |
| 16:36.28 | ChannelZ | if you scroll over to the end of it it might have a hostname or something of where it came from |
| 16:36.48 | n3hxs | When I looked at mine, it didn't make any sense.. |
| 16:37.03 | Katty | ya the very end says root@asterisk |
| 16:37.04 | ChannelZ | it might not |
| 16:37.07 | Katty | cheers. |
| 16:37.18 | Katty | ty for the help. i've set up my first ssh key! |
| 16:37.20 | Katty | happydance |
| 16:37.30 | ChannelZ | have fun. Time to go to work here. drat |
| 16:38.36 | Katty | ttyl |
| 16:42.29 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 16:42.44 | *** join/#asterisk sbszulu (~dundubala@41.18.170.26) |
| 16:45.02 | Katty | ok now that i've got my ssh key thingy setup |
| 16:45.13 | Katty | let's say i want to make copy.sh and cronjob that |
| 16:45.19 | Katty | how would i do a basic copy? |
| 16:45.34 | Katty | from boxa to boxb, using ssh |
| 16:45.55 | *** join/#asterisk drmessano-lt (~nonya@pdpc/supporter/active/drmessano) |
| 16:46.09 | Katty | hi danny |
| 16:48.05 | IsUp | it prompts for password |
| 16:48.10 | IsUp | i think its not possible Katty |
| 16:48.28 | Qwell | use key auth |
| 16:48.38 | Qwell | oh, you are. :p |
| 16:48.41 | IsUp | :p |
| 16:48.44 | Katty | well i've already setup the ssh key |
| 16:48.50 | Katty | so i don't think i'll be prompted for a password |
| 16:49.00 | Qwell | it'll prompt for the key password |
| 16:49.01 | IsUp | ah |
| 16:49.02 | Katty | but i don't really know how to copy from boxa to boxb |
| 16:49.06 | Qwell | scp |
| 16:49.08 | IsUp | use sshfs maybe? |
| 16:49.14 | IsUp | you can mount between servers |
| 16:49.31 | Katty | ok i could do a mount. that'd be fine |
| 16:49.31 | Qwell | scp srcfile user@server:~/destfile |
| 16:49.45 | IsUp | yeah you can use scp too |
| 16:49.46 | Qwell | or! |
| 16:49.48 | Katty | Qwell: will scp prompt me for the root password? |
| 16:49.57 | IsUp | no, since u have keys |
| 16:50.00 | Qwell | it'll do the same thing ssh would do |
| 16:50.05 | Katty | ok |
| 16:50.11 | Katty | let me make sure i understand this |
| 16:50.14 | Katty | cause i am /clueless/ |
| 16:50.16 | Qwell | (scp is literally a wrapper around ssh) |
| 16:50.18 | Katty | i'm on boxB. |
| 16:50.32 | Katty | i want to copy /path/to/file/onbox.a |
| 16:50.43 | Katty | to boxb, where i am /path/to/file/onbox.b |
| 16:51.02 | Qwell | scp user@boxa:/path/to/file/onbox.a /path/to/file/onbox.b |
| 16:51.18 | Qwell | it's just like cp, but you can add a user@host: |
| 16:51.24 | Qwell | (to either side) |
| 16:51.25 | Katty | tries it |
| 16:51.46 | espiceland | and it's encrypted. |
| 16:51.50 | IsUp | yeah, also you can use 'sshfs' to mount directories between servers, i was using it |
| 16:51.58 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:52.24 | Qwell | espiceland: true, that's a pretty important point |
| 16:52.52 | Katty | SQUEES |
| 16:52.58 | Katty | hugskwishes Qwell to bits |
| 16:54.17 | Katty | puts Qwell-bits back together to reform Qwell |
| 16:54.55 | Qwell | they don't go back together :( |
| 16:55.15 | Katty | pff |
| 16:55.26 | Katty | everything can be mended with hugs, or at least made better. |
| 16:55.32 | Katty | more tolerable |
| 16:55.35 | espiceland | Katty quelled Qwell! |
| 16:55.35 | WIMPy | Now we're left with Qmiserable |
| 16:57.45 | Katty | this seems like a much better way to copy files from my primary server to the backup system |
| 16:58.00 | Qwell | scp is awesome |
| 16:58.22 | WIMPy | rsync is even better |
| 16:58.42 | Qwell | no its not |
| 16:58.48 | *** join/#asterisk Benighted (~Benighted@209.119.153.210) |
| 16:58.57 | Benighted | Morning all |
| 16:59.09 | WIMPy | it is |
| 16:59.17 | kaii | -> religion? |
| 16:59.23 | IsUp | hello Benighted |
| 16:59.26 | nextime | for some things scp is better, for others rsync is |
| 16:59.42 | WIMPy | No, very usefull extra feature. |
| 17:00.06 | IsUp | bleeeeeeeeeeeehh |
| 17:00.21 | Katty | i'm just copying conf files from box a to box b |
| 17:00.26 | Katty | and i'll probably cron it at some point |
| 17:00.36 | Katty | i think scp is fine for that |
| 17:01.20 | Benighted | Im having an issue with FOP, I changed the reverse_transfer to 1 and did an amportal restart but still when transfering calls it hangs up on them. Google has been little help today. Got any suggestions? |
| 17:01.23 | Katty | Qwell's just one smart cookie. |
| 17:01.44 | Qwell | wonders what he was just tagged as |
| 17:01.54 | Katty | you should go look! |
| 17:01.57 | Qwell | oh |
| 17:02.04 | espiceland | You can even scp through an ssh tunnel to get through a firewall or into a box on a LAN. |
| 17:02.37 | Qwell | ssh is super fancy |
| 17:02.43 | Katty | you're super fancy. |
| 17:02.50 | Qwell | fact. |
| 17:03.23 | IsUp | Benighted: use #freepbx or #trixbox or #elastix whatever your release is |
| 17:03.24 | kaii | anyone has a use for a dialplan converter script which (almost) entirely converts 1.4 syntax to 1.6/1.8 syntax? (commas, execif, setcallerid, macro to gosub, ..) |
| 17:04.46 | *** join/#asterisk Jasnejac (~kvirc@81.91.106.59) |
| 17:06.56 | *** join/#asterisk luckman212 (~quassel@pool-173-77-253-145.nycmny.fios.verizon.net) |
| 17:12.15 | kaii | is there any way to test validity of a dialplan without executing (e.g. calling) every single piece of it ? |
| 17:12.56 | *** join/#asterisk omani (~hasan@33.37.69.80.in-addr.net-lab.net) |
| 17:13.40 | Naikrovek | how in the heck could i possibly have 10909 active sip channels |
| 17:13.42 | Naikrovek | ..... |
| 17:13.56 | omani | is it possible to have only one number at sipgate, and forward the caller to an extension by looking at the end of the one number which was called PLUS extension? |
| 17:14.06 | omani | ah man, kann einer deutsch? :( |
| 17:14.37 | kaii | Naikrovek: sip channels are also notifys, registers and subscriptions .. it doesnt mean "calls" |
| 17:14.45 | Naikrovek | yes i kow |
| 17:14.46 | Naikrovek | know* |
| 17:14.51 | Naikrovek | but ten thousand of them |
| 17:14.54 | *** join/#asterisk sbszulu (~dundubala@41.19.117.157) |
| 17:14.57 | Naikrovek | i only have about 100 phones |
| 17:15.23 | kaii | Naikrovek: which version? do you use realtime and qualify? are most of the channels type "OPTIONS"? |
| 17:15.39 | Naikrovek | they're not options |
| 17:15.43 | Naikrovek | they're 0x0 (nothing) |
| 17:15.50 | kaii | Naikrovek: sounds you hit the sip qualify / realtime / options flood bug |
| 17:16.00 | Naikrovek | not using realtime |
| 17:16.06 | kaii | Naikrovek: turn on sip debug and see whats going on |
| 17:16.34 | kaii | omani: nein das geht nicht (no this doesnt work) |
| 17:16.41 | kaii | omani: afaik sipgate does not support DID |
| 17:17.05 | omani | kaii: qry |
| 17:18.13 | kaii | dont see any |
| 17:18.14 | Naikrovek | sip debug is coming up empty. no genuine traffic right now so nothing going on |
| 17:18.22 | kaii | weird |
| 17:18.30 | kaii | which version of asterisk is that |
| 17:18.35 | IsUp | gotta go, gnite folks |
| 17:18.43 | Naikrovek | the active channels are people /trying/ to register to my box using IP spoofing, looks like. or it's a bug |
| 17:19.08 | Naikrovek | 1.4.22. yes i know it's old. yes i know it's buggy. |
| 17:20.57 | Naikrovek | upgrading is not an option at this time |
| 17:21.15 | kaii | do you see REGISTERS coming in ? |
| 17:21.25 | Naikrovek | yes |
| 17:21.29 | Naikrovek | well |
| 17:21.36 | Naikrovek | via sip show channels |
| 17:21.39 | Naikrovek | but not via sip debug |
| 17:22.04 | kaii | sounds like asterisk is not correctly terminating those dialogs |
| 17:22.10 | Naikrovek | no |
| 17:22.11 | kaii | is sip still working for you? |
| 17:22.16 | Naikrovek | yes the system is working fine |
| 17:22.30 | Naikrovek | this server was rebooted by some shit head in my other office 5 hours ago |
| 17:22.39 | Naikrovek | there is NO WAY that there have been that many registrations in that time |
| 17:22.45 | kaii | and you really really dont use realtime ? |
| 17:22.58 | Naikrovek | really really |
| 17:23.45 | Naikrovek | this server needs to be dumped. just wish i had the effin time to do it |
| 17:23.50 | kaii | you could do "database deltree SIP Registry" and then restart asterisk, but that would mean some downtime until all peers are registered again |
| 17:23.53 | Naikrovek | to migrate to asterisknow or something nodern |
| 17:24.11 | kaii | 1.4.22 really has some known bugs against sip |
| 17:24.52 | Naikrovek | this is why i rarely give out version numbers in IRC. I could be using the very latest up to date linux kernel, or one from 3 years ago. the answer is always "that one has known bugs" |
| 17:25.10 | Naikrovek | no matter what i say, it's always "dude that's your fault" |
| 17:25.15 | Naikrovek | that's not what YOU said |
| 17:25.37 | *** part/#asterisk espiceland (~erin@nat/digium/x-tknffcaguortuzyd) |
| 17:25.41 | Naikrovek | and you're correct in what you said |
| 17:25.49 | kaii | yes but that one has known bugs which cause exactly the behaviour you describe ;) |
| 17:25.51 | Naikrovek | but i knew what you were going to say before I even posed my question in here |
| 17:26.24 | kaii | cant help it .. there are two "solutions": do "database deltree SIP" and restart every time, OR upgrade |
| 17:26.37 | Naikrovek | i know.. |
| 17:27.02 | Linuturk | I'm reading up on how to support legacy polycom phones in my central config directory. http://supportdocs.polycom.com/PolycomService/support/global/documents/support/technical/products/voice/SoundPointIP_Phone_Support_in_SIP_3_2_TB35311.pdf My question is, if all the legacy phones are pointed to the 00000.cfg instead of the <mac>.cfg, as recommended in the article, how do I instruct the phone to use separate sip creds so they get different exten |
| 17:27.06 | Naikrovek | all its causing is a ton of register attempts from 127.0.0.1 |
| 17:27.07 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 17:27.48 | kaii | Linuturk: you need both .. to the global config in 000[..].cfg and the credentials in DEADBEEFMAC.cfg |
| 17:27.54 | kaii | *do |
| 17:28.22 | kaii | Linuturk: polycom phones first fetch 000000.cfg and then overwrite each setting which is defined in the mac specific cfg |
| 17:28.26 | Linuturk | kaii: the bulletin says to remove the separate mac address configs for all the extensions |
| 17:28.41 | Linuturk | legacy extensions* |
| 17:29.40 | Linuturk | it specifically says that if you leave the <mac>.cfg on the server, they will be used instead of the 0.cfg |
| 17:31.05 | kaii | well then .. i dont know |
| 17:31.44 | kaii | do firmware updates or something .. this sounds wrong, cuz as you said: how do you setup individual credentials then? |
| 17:32.22 | kaii | Naikrovek: from 127.0.0.1 ?? weird. then go hunt the local sip client .. maybe in sip.conf some stupid set up asterisk to register against itself ? |
| 17:32.41 | Linuturk | well, right now, I've got a <mac>.cfg for each phone, that points it to e1000.cfg (extension 1000) that tells it to register to asterisk as extension 1000 |
| 17:32.59 | Linuturk | as an example |
| 17:33.55 | Linuturk | is there a way to tell asterisk to use a certain mac address as an extension? that just doesn't seem right |
| 17:35.37 | kaii | Linuturk: ya well .. polycom sucks at this. |
| 17:35.53 | kaii | Linuturk: and no, there is no workaround for that in asterisk |
| 17:36.19 | kaii | you would have to do these modifications in each <mac>.cfg and not use 000...cfg at all |
| 17:36.43 | kaii | this is what the doc says |
| 17:37.00 | kaii | or: configure credentials on the device by hand and not provisioning |
| 17:38.18 | Naikrovek | ... you tell the polycom which extension it shoudl be via the <mac>.cfg file |
| 17:39.37 | Naikrovek | which legacy phones are you talking about |
| 17:40.03 | Linuturk | Naikrovek: I've mostly got a bunch of 300 and 301's |
| 17:40.25 | Linuturk | a few 600's I'm replacing with 650's |
| 17:40.30 | Linuturk | hence this problem |
| 17:40.38 | *** join/#asterisk Defraz (~Defraz@207.109.45.74) |
| 17:40.41 | Linuturk | I can't provision the 650's on the old firmware |
| 17:42.29 | Naikrovek | so have separate firmware? i do that and it works well |
| 17:43.03 | Naikrovek | i have DHCP reservations for every device in my network, and when I want a phone to get a non-standard config, I change it's DHCP setting to pull config from a different user on my FTP server. |
| 17:43.17 | Naikrovek | s/it's/its/ |
| 17:43.59 | Linuturk | that's an interesting approach |
| 17:44.13 | Linuturk | all my phones are in a separate vlan |
| 17:44.33 | Linuturk | so, dhcp reservations weren't really neaded |
| 17:44.33 | nightwalk | Are there any other flags or anything I can use to get asterisk to tell me why it's voluntarily shutting back down on startup? No combination of -v/-vvv, -c, or -g works :/ |
| 17:44.37 | Linuturk | needed* |
| 17:45.20 | Naikrovek | well you can set up DHCP reservations for the new phones and point them to a new config somewhere separate from the legacy config |
| 17:45.36 | Naikrovek | if you have DHCP on that vlan |
| 17:45.38 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
| 17:45.48 | Naikrovek | <-- doesn't have any vlan experience, nor DHCP forwarding experience |
| 17:46.26 | *** join/#asterisk grkblood (967da669@gateway/web/freenode/ip.150.125.166.105) |
| 17:47.48 | grkblood | is there any type of softphone or conference manager that allows people to call in to a conference you're hosting and adjust the volumes of the participants individually? |
| 17:48.02 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 17:52.45 | Naikrovek | you wan tto let the manager adjust the volume of the participant's phones? or do you want to mute |
| 17:56.00 | *** join/#asterisk JamesHarrison (~jharr@hometree.mmmetrics.co.uk) |
| 17:56.37 | grkblood | Naikrovek: adjust and mute |
| 17:57.09 | grkblood | i want the person hosting the conference to have the ability to turn peopels volume up if other peopel have trouble hearing them |
| 18:01.24 | grkblood | ive searched and searched and cant find anything |
| 18:03.20 | Naikrovek | why don't you trust people to adjust their own volume |
| 18:03.47 | Naikrovek | and how will the person running the conference knwo if someone can't hear him? |
| 18:04.04 | Naikrovek | if you're going to say something that you don't want everyone to hear, say it at another time |
| 18:05.45 | Brixius | grkblood: there is no volume control mixer board for conferences that I'm aware of. |
| 18:05.57 | Naikrovek | i can't even reason why you would want one |
| 18:05.59 | grkblood | let me rephrase that |
| 18:06.08 | grkblood | if the host thinks their volume needs to be higher |
| 18:06.19 | grkblood | Naikrovek: for a radio show |
| 18:06.35 | grkblood | if i have a live radio show and im usign a softphone to have two peopel on at once as guests |
| 18:06.41 | grkblood | i want to level there volumes out |
| 18:06.51 | Naikrovek | a radio show would have an engineer that would adjust the levels BEFORE the audio stream went into the phone system |
| 18:07.02 | grkblood | right, but there is no adjusting |
| 18:07.05 | grkblood | its not even possible |
| 18:07.09 | Naikrovek | not in the phone system, no |
| 18:07.12 | Naikrovek | you adjust it BEFORE |
| 18:07.14 | grkblood | thats what im asking |
| 18:07.27 | Brixius | grkblood: For a radio show, I would have several phones, each patched into your engineers board. |
| 18:07.31 | Naikrovek | you speak into a mic, this goes into a sound board. a soundboard output goes into the phone system |
| 18:07.53 | Naikrovek | this way you can control levels BEFORE the phone system gets involved |
| 18:07.59 | grkblood | Brixius: or a phone with multiple lines |
| 18:08.14 | Naikrovek | that's why you can't find a solution; you're going about it the wrong way i think |
| 18:08.21 | Brixius | a phone with multipul lines will have a single audio out. |
| 18:08.34 | grkblood | i know, i need a phone with a line mixer |
| 18:08.41 | bmoraca_work | how do i make asterisk ignore P-Asserted-Identity? |
| 18:08.41 | Naikrovek | you need a mixer AND a phone |
| 18:08.58 | bmoraca_work | or perform routing not based on incoming callerid and IP only? |
| 18:08.59 | grkblood | i need a mixer IN the phone |
| 18:09.11 | Naikrovek | good luck finding one |
| 18:09.19 | Naikrovek | they dont' exist because that's not how you do it |
| 18:09.33 | grkblood | well, softphones arnt the answer then |
| 18:09.47 | grkblood | because you would need multiple computers to adjust multiple volumes |
| 18:09.47 | nightwalk | grkblood: I think what they're telling you is that you might want to look at chan_oss |
| 18:10.01 | Naikrovek | "google is stupid! I can't find a car with the driver's seat facing backwards! stupid google!" <-- example of how you know you're doing something wrong |
| 18:10.29 | Brixius | grkblood: I think your best bet for that kind of thing using off the shelf methods, would be to run several softphones in windows and then use the windows mixer for adjusting the volumes of each softphone. |
| 18:10.39 | grkblood | Naikrovek: im trying to explain to you why multple soft phones isnt really an option for me |
| 18:10.50 | Naikrovek | i'm not telling you to use multiple softphones |
| 18:11.10 | grkblood | you said multiple phones did you not? |
| 18:11.17 | Naikrovek | A) you can't adjust the volume of other participant's phones without modifying Asterisk itself. |
| 18:11.22 | Brixius | no, he didn't I did. |
| 18:11.28 | Naikrovek | B) you don't want to adjust other people's phones. |
| 18:11.39 | grkblood | i dont want to adjust their phone |
| 18:11.44 | Naikrovek | C) if you want to adjust your OWN volume, then you adjust it BEFORE it goes into the softphone |
| 18:11.49 | grkblood | i want to adjust it locally on my phone |
| 18:11.49 | bmoraca_work | asterisk's peer selection for inbound calls is really stupid. |
| 18:12.56 | grkblood | Naikrovek: i get that, but if youre not saying get multiple phones then how do you go about adjust each lines volume before you pipe it in |
| 18:13.10 | Naikrovek | why the hell do you need multiple phones |
| 18:13.19 | Naikrovek | you're assuming i know something that i don't |
| 18:13.24 | Naikrovek | about your situation, i mean |
| 18:13.33 | grkblood | Brixius: you get what im saying right? |
| 18:14.24 | Naikrovek | you want to adjust the call volume of each caller ? |
| 18:14.29 | Brixius | grkblook: I understand you want to have dialin participants to your radio show, then be able to adjust each participants output volume to your show if it's quiet. |
| 18:14.40 | Naikrovek | you want to adjust what you hear from them |
| 18:14.47 | Naikrovek | okay |
| 18:14.51 | grkblood | yes Naikrovek |
| 18:15.01 | grkblood | thats what ive been saying, lol |
| 18:15.02 | grkblood | :) |
| 18:15.11 | Naikrovek | i have no idea how to do that with asterisk or a sofphone |
| 18:15.15 | *** part/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 18:15.29 | Naikrovek | i don't think you can do that |
| 18:15.38 | grkblood | Brixius: the problem is a lot of softphone ive dealt with control the master volume |
| 18:15.41 | grkblood | like x-lite |
| 18:16.01 | grkblood | what softphone would you suggest? |
| 18:16.12 | Naikrovek | you'll have to put each caller into a softphone or some pbx to accept the call and handle SIP and RTP, then pipe that to a mixer somewhere, then master that into yoru show audio, then the same for reverse |
| 18:16.51 | grkblood | yea, if they just made a softphone with what i wanted this would be so much easier :) |
| 18:17.02 | grkblood | someone needs to make one |
| 18:17.07 | Naikrovek | no |
| 18:17.16 | ChannelZ | Don't you have a mixer for this type of thing? |
| 18:17.26 | grkblood | oh lord.... |
| 18:17.29 | Naikrovek | it is not the job of the softphone to apply effects (such as gain adjustment) that is the job of the mixer |
| 18:17.31 | ChannelZ | Preferably with a compressor/limiter? |
| 18:17.33 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 18:17.34 | bmoraca_work | Can anyone look at a pastebin and tell me why Asterisk is choosing the peer it is, and not the peer that matches the source IP of the request? http://pastebin.com/jGcFSAsp |
| 18:17.46 | Brixius | I don't know, but I would not use the volume control of the softphone, but rather open up like windows 7's volume control where it lists each app's sound output. so you can adjust each apps sound. If that doesn't work, I think your best bet is to throw together some custom client(Find a sip phone library and hack somehting together). |
| 18:17.50 | Naikrovek | bmoraca_work: quiet the grownups are arguing |
| 18:17.51 | Naikrovek | :) |
| 18:18.14 | grkblood | Brixius: ive heard of that |
| 18:18.23 | grkblood | maybe i should upgrade to win 7 |
| 18:18.28 | Naikrovek | that's exactly what i've been suggesting |
| 18:18.35 | Brixius | I think vista has it as well |
| 18:18.38 | Naikrovek | you need to put a mixer in the setup here somewhere |
| 18:18.46 | Naikrovek | linux has mixers too |
| 18:18.59 | Brixius | I'm would be supprised if linux didn't have the same functionality. |
| 18:19.10 | ChannelZ | bmoraca_work: without seeing your sip.conf your guess is better than anyone else's here why it's matching what it's matching |
| 18:19.24 | grkblood | ok, so i think this is what yall are saying, run like 4 softphones on 1 pc, route the audio from each phone into a virtual mixer |
| 18:19.31 | Naikrovek | yes |
| 18:19.43 | Naikrovek | then the output of the mixer will be half of your show audio |
| 18:19.46 | Naikrovek | the other half will be you |
| 18:20.06 | Brixius | Otherwise you could put multiple sound devices in your computer and have each softphone output to a seperate device and put that into your mixer(beter yet, get a professional multi channel sound output board to route the sound out of the pc) |
| 18:20.08 | Naikrovek | linux has a system for this exact thing but i dunno what it's calld right now |
| 18:20.20 | grkblood | Naikrovek: ive looked into that, havent foudn a mixer capable of accepting other progs audio |
| 18:20.21 | grkblood | rewire can do that but thats for midi stuff i think |
| 18:20.30 | Naikrovek | there is one, it's new |
| 18:20.36 | grkblood | Naikrovek: you talkin about Jackd |
| 18:20.36 | bmoraca_work | ChannelZ: which portions of SIP.conf do you need? the peer that should be matched is called "pwsip" and is using IP authentication, and the peer it is matching is another peer that is being registered to by another device. i can pastebin the [general] section as well |
| 18:20.39 | Naikrovek | it creates virtual audio devices and pipes audio between them |
| 18:20.44 | Naikrovek | i don't know the name of it |
| 18:21.20 | ChannelZ | bmoraca_work: both peers |
| 18:21.23 | grkblood | theres something on linux called JACK that does somethign like that |
| 18:21.31 | bmoraca_work | ChannelZ: i think the culprit might be "match_auth_username=yes", but I need to have that because Asterisk has issues matching peers properly. |
| 18:24.41 | *** join/#asterisk eject_ck (~eject_ck@83-218-246-156.dynamic.vega-ua.net) |
| 18:30.12 | *** join/#asterisk JonnyD_work (~Jon@12.222.63.34) |
| 18:30.37 | *** join/#asterisk sequencer (~something@196.218.255.29) |
| 18:30.41 | sequencer | hi all |
| 18:31.01 | sequencer | any one knows where to find a call log terminology? |
| 18:31.10 | sequencer | what does zombie means, and so on.. ? |
| 18:31.14 | JonnyD_work | does anyone know how rrmemory decides who gets the first call? |
| 18:31.36 | JonnyD_work | and is there a way to modify it? |
| 18:32.08 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:33.10 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 18:33.34 | bmoraca_work | ChannelZ: here are the two peers: http://pastebin.com/YGR9Cizt |
| 18:34.00 | bmoraca_work | ChannelZ: calls work properly on the pwsip trunk, as long as my callerid does not match one of my other existing peers. |
| 18:34.02 | ChannelZ | I think a zombie is just when a call has been disconnected but * didn't really know of it (didn't get a BYE, etc) |
| 18:34.53 | sequencer | ChannelZ thanks, i was looking into a terminology for the logs, is there anywhere i can find ( what does the word... means? ) |
| 18:35.36 | sequencer | also, am getting alot of dropped calls due to fax detection, has anyone had similar issues? |
| 18:35.43 | *** join/#asterisk legis (~legis@unaffiliated/legis) |
| 18:36.59 | ChannelZ | I haven't. Do you use fax detection normally? |
| 18:37.11 | sequencer | yes, am using it on all extensions |
| 18:37.27 | sequencer | so that people can send faxes to DIDs directly |
| 18:37.33 | ChannelZ | hmm. Do you phone Mariah Carey a lot? |
| 18:37.41 | sequencer | sometimes |
| 18:37.53 | ChannelZ | that damn dolphin voice of hers |
| 18:38.01 | sequencer | lol |
| 18:38.04 | legis | Hi, I need a SFP box for asterisk with a TDM410P or AEX410 card, any recommendations? |
| 18:38.27 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 18:40.42 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 18:40.50 | ChannelZ | bmoraca_work: do both those peers come from the same IP? |
| 18:41.00 | bmoraca_work | ChannelZ: no they do not |
| 18:41.04 | Qwell | legis: SFP? |
| 18:41.21 | bmoraca_work | ChannelZ: but I have other peers which do come from the same IP, hence the need for "match_auth_username=yes" |
| 18:41.54 | legis | Qwell: sorry, SFF |
| 18:42.00 | legis | small form factor |
| 18:42.09 | ChannelZ | but you're saying the two that you pasted are the two getting confused. |
| 18:42.43 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 18:42.49 | bmoraca_work | ChannelZ: they all get confused if a call comes in over the "pwsip" peer with a caller id that matches the name of one of my other peers |
| 18:43.05 | bmoraca_work | ChannelZ: interestingly enough, i have another peer set up identically that does not have this problem |
| 18:43.13 | bmoraca_work | the only difference i can see is P-Asserted-Identity |
| 18:43.39 | ChannelZ | bmoraca_work: from sip.conf: http://pastebin.com/MCr0Wgzc |
| 18:43.57 | ChannelZ | You'll have to sort out all your peers since we're only seeing a small picture |
| 18:44.58 | bmoraca_work | ChannelZ: that doesn't apply with "match_auth_username=yes", at least from the point of view of calls coming in on peers that are not using IP-based authentication |
| 18:46.26 | ChannelZ | the two peers you posted do not have such a config. One doesn't even specify the host, dynamic or otherwise (maybe it's being inherited from your general config) So I don't know what the heck you have going on |
| 18:46.33 | drift- | is it possible to get free outgoing callls voip |
| 18:46.36 | bmoraca_work | match_auth_username=yes is a general config |
| 18:47.02 | *** join/#asterisk jmurph05 (~jmurph05@jose-tc.ctc.biz) |
| 18:47.12 | bmoraca_work | ChannelZ: as I said, there is another identical peer that does not have this problem for inbound calls. |
| 18:47.52 | *** join/#asterisk dajhorn (~dajhorn@li211-132.members.linode.com) |
| 18:48.19 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 18:50.50 | drift- | is it possible to pick up the phone and use google voice to call out? |
| 18:51.02 | drift- | astersik + google voice... |
| 18:51.14 | drift- | get free outgoing calls has anyone done that here yet? |
| 18:51.26 | Qwell | sure |
| 18:51.37 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 18:51.51 | Qwell | https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 18:52.13 | eject_ck | Guys I have problem with my ISP - I have server in colocation and I figured that I can't register on internetcalls server with asterisk. |
| 18:52.28 | jmurph05 | I keep getting a dependency error for dahdi. it says a need a different kernel, but i have that exact kernel running |
| 18:52.30 | *** join/#asterisk txwikinger (~quassel@sblug/member/txwikinger) |
| 18:52.42 | Qwell | jmurph05: it needs the source for that kernel |
| 18:52.54 | jmurph05 | Qwell: how do i set that? |
| 18:53.04 | Qwell | what distro? |
| 18:53.11 | jmurph05 | CentOS 5.5 |
| 18:53.13 | eject_ck | I've asked ISP if 5060/udp blocked and thay said no . |
| 18:53.26 | Qwell | jmurph05: yum install kernel-devel-`uname -r` |
| 18:53.32 | fenrus | both isps? |
| 18:53.55 | fenrus | guess you have one isp at the colo, and one isp at home/office ? |
| 18:54.33 | eject_ck | I've created sip extension and I see that I'm able to register on that server with SIP client from external host (outside colocation), but register => still not work |
| 18:54.46 | jmurph05 | Qwell: thanks |
| 18:54.50 | eject_ck | I see outgoing packets but no traffic on target server |
| 18:55.08 | eject_ck | When I changed port in register to 5061 it appeared |
| 18:55.15 | eject_ck | something misterious |
| 18:55.40 | jmurph05 | Qwell: it still says i'm missing the dependency |
| 18:55.42 | fenrus | did you ask BOTH your colo isp and your other-site ISP about firewalls ? |
| 18:55.52 | eject_ck | fenrus: yes |
| 18:55.53 | Qwell | What, exactly, does it say? |
| 18:55.55 | fenrus | it sounds like they have something running. |
| 18:56.06 | fenrus | or your own router/firewall's eating the packets |
| 18:56.37 | eject_ck | fenrus: I have working asterisk on Server1 (not problematic one), and Server2 in colocation |
| 18:57.11 | eject_ck | I've set registration from Server2 to Server1 and vice versa on asterisks. |
| 18:57.20 | jmurph05 | Qwell: "missing dependency: kernel-x86_64 = 2.6.18-194.32.1.e15 is needed by package kmod-dahdi-linux-2.4.0" |
| 18:57.28 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v010-133.mobile.uci.edu) |
| 18:57.30 | fenrus | why use SIP between the servers? :) |
| 18:57.34 | Qwell | and what distro is this? |
| 18:57.43 | eject_ck | No traffic on port 5060 in from Server1 on Server2 and no in traffic on Server2 from Server1 |
| 18:57.46 | jmurph05 | latest CentOS |
| 18:57.53 | fenrus | id try IAX instead |
| 18:58.05 | Qwell | and `rpm -q kernel` says its installed? |
| 18:58.18 | eject_ck | fenrus: original problem that I can't register on my sip provider (internetcalls) |
| 18:58.23 | jmurph05 | Qwell: yes |
| 18:58.27 | eject_ck | fenrus: original problem that I can't register on my sip provider (internetcalls) on Server2 |
| 18:58.35 | fenrus | ah, okay |
| 18:58.37 | Qwell | How are you trying to install it? |
| 18:59.30 | eject_ck | fenrus: biggest mistery that I'm able to register with X-lite from Server1 network on Server2 |
| 18:59.55 | jmurph05 | Qwell: it installed via yum. rpm -q tells me "kernel -2.6.18-194.e15" without the '32.1' |
| 19:00.07 | Qwell | Then you aren't running the kernel it requires. |
| 19:00.07 | eject_ck | So I can guess that problem is with asterisk client :( |
| 19:00.25 | fenrus | eject_ck, i'd try removing all config regarding the sip peering in both servers and redo it |
| 19:00.37 | fenrus | using as little options as possible to begin with |
| 19:00.43 | bmoraca_work | the only difference in the invite from my as5400 (which matches peers correctly) and the SIP service provider (which doesn't) is P-Asserted-Identity... |
| 19:00.48 | jmurph05 | Qwell: yum install kernel-devel with the required one? |
| 19:00.48 | bmoraca_work | wonder if i can get them to turn that off |
| 19:00.56 | Qwell | jmurph05: You need to enable the updates repo from CentOS. That is the kernel it shipped with, and is very old. |
| 19:01.08 | eject_ck | fenrus: just tried to build it from source with default configs (just added register line with correct ip and extension) |
| 19:01.11 | eject_ck | same result |
| 19:01.48 | eject_ck | what's difference between X-lite and Asterisk as SIP client? Same 5060/udp |
| 19:03.17 | fenrus | try asterisk on another machine ? |
| 19:03.26 | jmurph05 | Qwell: trying an update now. thanks |
| 19:04.29 | eject_ck | ok, I've found difference : ) asterisk uses port 5060 as src port |
| 19:04.36 | eject_ck | And it seams it |
| 19:04.59 | eject_ck | And it seams it's really blocked at ISP on Server1 for incoming traffic |
| 19:05.33 | eject_ck | Can I change port what asterisk uses for outgoing sip registration ? |
| 19:05.52 | fenrus | port= i think |
| 19:05.59 | fenrus | hm, |
| 19:06.04 | fenrus | thats the destinationport :D |
| 19:06.41 | Katty | hai fenrus! |
| 19:07.16 | fenrus | Katty, oh hai! |
| 19:09.17 | eject_ck | sec |
| 19:09.36 | eject_ck | any way to change outgoing port ? |
| 19:09.56 | fenrus | hm, dont know if the 'bindport' satement gives you anything |
| 19:12.15 | eject_ck | let me see |
| 19:13.56 | *** join/#asterisk Brimmy (~Brimmy@142.150.115.43) |
| 19:14.34 | eject_ck | wow! Works! |
| 19:17.16 | eject_ck | I've changed udpbindaddr=xxx.xxx.xxx.xxx:5061 |
| 19:17.20 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 19:17.46 | eject_ck | a bit problematic for me to explain what's blocked on ISP side |
| 19:23.33 | fenrus | =) |
| 19:27.22 | eject_ck | seems like they block if src and dst ports are the same |
| 19:29.28 | fenrus | theres no nat'ing router between that eats the packets? :) |
| 19:29.59 | eject_ck | no nat |
| 19:30.07 | eject_ck | Wow! I'm right! |
| 19:30.18 | fenrus | odd thing to do |
| 19:31.01 | eject_ck | I've changed udpbindaddress=xxx.xxx.xxx:5061 and changed register => xxx:xxx@myip:5061/xxx and see that they blocks it |
| 19:31.16 | eject_ck | Can anybody explain why it make sense ? |
| 19:31.44 | eject_ck | from ISP's point of view |
| 19:32.12 | fenrus | no, |
| 19:32.17 | fenrus | since it does not make sense |
| 19:32.33 | *** join/#asterisk Corydon76-home (orange@c-69-137-80-31.hsd1.tn.comcast.net) |
| 19:32.33 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 19:32.36 | fenrus | i think that this is some kind of application problem |
| 19:33.43 | eject_ck | fenrus: what you mean ? |
| 19:34.08 | eject_ck | what's problem in next communication example |
| 19:34.27 | eject_ck | 21:20:48.711535 IP xxx.xxx.xxx.xxx.5060 > 94.75.247.45.5060: SIP, length: 404 |
| 19:34.35 | eject_ck | why ISP block it ? |
| 19:35.03 | fenrus | an isp could block SIP to force you use their service |
| 19:35.28 | fenrus | but this port based scheme sounds really far fetched |
| 19:35.48 | fenrus | never heard of anything like this before |
| 19:36.02 | fenrus | and i work for an quite large ISP |
| 19:41.06 | *** join/#asterisk jetlag (~jetlag@pool-173-61-239-155.cmdnnj.east.verizon.net) |
| 19:44.32 | *** join/#asterisk sequencer (~something@196.218.255.29) |
| 19:44.37 | sequencer | hi all |
| 19:45.06 | sequencer | does anyone know how to setup up voicemail indicator to show when theres a voicemail on a polycom phone? |
| 19:45.32 | sequencer | i know this function exists on polycom phones but i dont know how to set it up on * |
| 19:46.30 | citywok | sequencer: mailbox=mailbox#@context in sip.conf |
| 19:46.42 | eject_ck | fenrus: they stated that VoIP not forbidden and more they stated that there is no any blocks on their routers |
| 19:47.14 | sequencer | citywok i have it as mailbox=ext@device |
| 19:47.25 | sequencer | is that in * or in polycom settings? |
| 19:47.47 | Naikrovek | asterisk |
| 19:48.09 | citywok | sequencer: sip.conf in asterisk, and ext is fine if ext = mailbox :P |
| 19:48.35 | sequencer | citywok for instance 101@device ? |
| 19:48.52 | citywok | i use the default context for voicemail, so i would do 101@default |
| 19:49.19 | sequencer | am using freepbx though for easier management |
| 19:49.38 | eject_ck | Btw, that's in quite cool DC in Kiev, Ukraine owned by big Ukrainian telco company GoldenTelecom (now owned by Russian Vimpelcom NYSE:VIP) |
| 19:50.04 | Qwell | mmm, kiev. tasty |
| 19:50.49 | eject_ck | I've spent 1week after submiting ticket and they just resolved it :) |
| 20:03.12 | *** join/#asterisk ariel_ (~chatzilla@173-107-183-222.pools.spcsdns.net) |
| 20:09.35 | *** join/#asterisk iscario (~quassel@31.248.101-84.rev.gaoland.net) |
| 20:13.05 | n3hxs | eject_ck We have a ticket open with Verizon that is 9 months old. |
| 20:13.24 | eject_ck | n3hxs: the same issue ? |
| 20:14.10 | eject_ck | n3hxs: I have solution for this problem to bind to :5061 on localserver, but I have to change configs on all clients |
| 20:14.12 | n3hxs | yeah, they found the issue which is a noisy trunk from Level 3 and busied it out. so it is sorta fixed till some tech says " why is this busied out. |
| 20:15.08 | n3hxs | Yeah patches to fix non issues ;) |
| 20:15.14 | eject_ck | n3hxs: fun, as far I know my ISP uses Level 3 carrier links to US and Europe |
| 20:16.26 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.132.25) |
| 20:20.04 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 20:26.37 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 20:27.58 | JonnyD_work | does anyone know how a queue in rrmemory decides what dynamic agent get the first call that comes into the queue? |
| 20:34.55 | *** join/#asterisk bio-tty (~c@153.182.251.212.customer.cdi.no) |
| 20:36.25 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 20:43.48 | Corydon76-home | JonnyD_work: first in the list |
| 20:44.08 | JonnyD_work | so the first person who logs into the que? |
| 20:44.26 | Corydon76-home | which may be the last person who logs into the queue, depending upon how it was coded. |
| 20:44.58 | Corydon76-home | but there's no deterministic intent for that |
| 20:45.32 | Corydon76-home | It's random amongst all the people who have yet to receive a call since logging in |
| 20:46.31 | Corydon76-home | THAT comes down to an implementation detail |
| 20:46.53 | JonnyD_work | can i get it to try the first person who logged in |
| 20:47.05 | JonnyD_work | is there a way to tell the queue to do that? |
| 20:47.19 | Corydon76-home | No |
| 20:47.36 | Corydon76-home | not without muddying yourself in the code |
| 20:47.37 | JonnyD_work | okay thats what i though |
| 20:48.18 | JonnyD_work | when you say first in the list do you mean the list that shows up when doing a show queues? and is it from top to bottom? |
| 20:48.22 | Corydon76-home | Certain versions MAY be the first person or the last person, who logged in. |
| 20:48.36 | JonnyD_work | i am using 1.4.31 |
| 20:48.39 | Corydon76-home | First in the internal list |
| 20:49.22 | Corydon76-home | Yes, first in the list, when you use the command line, since it's the same search operation |
| 20:49.57 | JonnyD_work | so its pretty much a crap shoot? my client is complaining that the lowest number extension is getting the first call |
| 20:50.23 | Corydon76-home | Yep, it's effectively random |
| 20:50.23 | JonnyD_work | i have yet to verify that since i am not going to waste my time looking though logs to disprove him |
| 20:50.48 | JonnyD_work | yeah that's anoying i like when things are predictable |
| 20:51.01 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 20:51.27 | Corydon76-home | They are predictable, if you want to get down in the details. Most people don't, and so for them, it's considered random. |
| 20:51.43 | *** part/#asterisk jmurph05 (~jmurph05@jose-tc.ctc.biz) |
| 20:52.29 | JonnyD_work | well if i have to go in and modify the code to get it todo what the client wants he is out of luck |
| 20:54.59 | De_Mon | well that was anticlimactic. I was having problem authenticating to mysql 5.1 via ODBC and it turned out I had to supply the user/pass in odbc.ini instead of using the commandline. |
| 21:07.44 | *** join/#asterisk delroy (~delroy@tba.usask.ca) |
| 21:08.41 | delroy | Anyone ever have problems with DTMF tones and stuttering? using analog Sangoma PCIe card andAastra handsets |
| 21:10.03 | delroy | using inband on the phone may solve it, but then the comedian voicemail app doesnt see any tones when it is accessed |
| 21:20.29 | *** join/#asterisk DJClean (~djclean@unaffiliated/djclean) |
| 21:22.07 | *** join/#asterisk geneg1 (~gene@173-230-163-176.cable.teksavvy.com) |
| 21:28.46 | *** join/#asterisk alphawave (~aw@unaffiliated/alphawave) |
| 21:33.12 | *** join/#asterisk eject_ck (~eject_ck@83-218-246-156.dynamic.vega-ua.net) |
| 21:39.26 | *** join/#asterisk ningia (~gain@host126-91-dynamic.6-79-r.retail.telecomitalia.it) |
| 21:40.54 | Katty | thwaps drmessano-lt |
| 21:41.52 | citywok | i've got a 1.8.1.1 system that's crashed a couple times on a call transfer that came in to a queue, anybody else had similar problems? |
| 21:43.14 | drmessano-lt | lol |
| 21:43.16 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 21:43.22 | drmessano-lt | is full |
| 21:43.39 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:49.11 | citywok | drmessano: mean, don't laugh at my crashes :P |
| 21:52.38 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 21:58.01 | *** join/#asterisk Gadfly (~Gadfly@cpe-204-210-139-235.hvc.res.rr.com) |
| 22:05.01 | *** join/#asterisk grkblood13 (~grkblood1@c-68-58-177-90.hsd1.sc.comcast.net) |
| 22:06.00 | grkblood13 | anyone ever launch the same softphone multiple time on XP? if so what phone and how did you do it |
| 22:27.39 | bmoraca_work | is there any way i can make Asterisk match peer based on IP first, rather than the From: header? |
| 22:29.33 | WIMPy | type=peer should do that |
| 22:29.38 | bmoraca_work | you'd think so, but no |
| 22:29.43 | bmoraca_work | actually, it's weird |
| 22:29.54 | bmoraca_work | i have TWO peers set up, one of them works properly, the other doesn't |
| 22:30.07 | bmoraca_work | one matches properly, the other matches based on From: |
| 22:30.42 | bmoraca_work | the only difference between the INVITEs is that the one that doesn't work uses P-Asserted-Identity |
| 22:30.51 | bmoraca_work | but, from what i've seen, Asterisk doesn't use that |
| 22:31.01 | bmoraca_work | i'm at a loss...can't figure it out |
| 22:31.21 | WIMPy | makes a mental note that that was something to be tested. |
| 22:36.06 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 22:38.16 | *** join/#asterisk Defraz (~Defraz@207.109.45.74) |
| 22:42.37 | *** join/#asterisk nix8n82 (~nate@63.162.28.112) |
| 22:43.20 | grkblood13 | anyone ever use sflphone? |
| 22:45.08 | *** join/#asterisk leejohn (~stealth@58.69.15.221) |
| 22:45.36 | leejohn | hi! recent commit to svn 1.8 branch fails to compile |
| 22:46.00 | leejohn | chan_dahdi.c: In function setup_dahdi_int: |
| 22:46.00 | leejohn | chan_dahdi.c:17656: error: incompatible types when assigning to type struct dahdi_chan_conf * from type struct dahdi_chan_conf |
| 22:46.13 | russellb | well that's no good |
| 22:46.22 | russellb | Qwell: ^^^^ |
| 22:46.25 | russellb | that was you :-p |
| 22:46.49 | leejohn | i already file a bug report and i go directly here :D |
| 22:47.29 | russellb | our automated builds caught it, too :-) |
| 22:47.30 | russellb | http://bamboo.asterisk.org/browse/AST18-LUCID-AMD64-185/log;jsessionid=1ipxk73p2r6g |
| 22:49.12 | leejohn | chilax, i'll just wait for this to be fixed thanks guys |
| 22:50.59 | Qwell | wtf. that didn't even give a merge conflict. |
| 22:51.27 | russellb | heh |
| 23:00.01 | drmessano | Is it fixed yet? |
| 23:01.23 | *** join/#asterisk saxa (~sasa@host242-95-static.223-217-b.business.telecomitalia.it) |
| 23:01.44 | leejohn | drmessano: not yet |
| 23:02.19 | saxa | hello, any of you knows how to put the correct extension number for voicemail to work with grandstream gxp2120 voip phone mailbox button ? |
| 23:04.12 | drmessano | Luckily, Qwell has 26 mins before he's the guy stuck in the building after-hours fixing a commit |
| 23:04.27 | drmessano | We call that motivation |
| 23:05.03 | leejohn | drmessano: :D |
| 23:06.07 | leejohn | saxa: sorry i don't have GS phone here at my installation |
| 23:12.52 | Qwell | should compile again |
| 23:13.38 | drmessano | With 18 mins to spare |
| 23:13.40 | drmessano | Well done |
| 23:15.38 | Qwell | motivation: I've got some beer to drink. |
| 23:15.49 | russellb | Qwell: you cheated |
| 23:15.51 | Qwell | really though, I just reverted it for now, until I could test something. |
| 23:15.55 | Qwell | QUIET YOU |
| 23:15.56 | russellb | :-p |
| 23:16.14 | Qwell | I want to make sure this is right, before my changes. 1.8 had to go and be all different. pfft. |
| 23:16.23 | russellb | nods |
| 23:16.38 | Qwell | it's silly how it all works, heh |
| 23:16.42 | Qwell | then, now, always |
| 23:18.34 | russellb | and on that note. |
| 23:18.43 | russellb | bails |
| 23:20.44 | bio-tty | dundi and enum cannot be used to call pstn numbers to be billed right? |
| 23:21.03 | bio-tty | i mean, billing is completely out of scope, right? |
| 23:24.46 | saxa | leejohn: thx, its strange because in some way asterisk says to the phone that there are voice messages, but when I press the mailbox button nothing happens, it does not dial the voicemail. |
| 23:24.51 | bio-tty | i guess it is for target numbers that can be reached on Internet without entering psnt to his thus numbered landline. |
| 23:25.14 | bio-tty | or maybe for landlines that can locally be reached for free? |
| 23:25.49 | WIMPy | saxa: Did you set vmexten? |
| 23:25.58 | saxa | yes |
| 23:26.10 | saxa | WIMPy: you mean in the phone ? |
| 23:26.25 | WIMPy | no, sip.conf |
| 23:26.30 | saxa | yes |
| 23:26.40 | saxa | it is 1001@default |
| 23:27.58 | saxa | yes, mailbox=1001@default |
| 23:28.27 | WIMPy | I said 'vmexten', not 'mailbox'. |
| 23:28.43 | saxa | so in my extensions.conf I have an exten => 900,Voicemail() |
| 23:28.49 | saxa | WIMPy: then its no |
| 23:29.04 | WIMPy | Then do it. |
| 23:29.05 | saxa | let me try |
| 23:29.14 | bio-tty | so, aparently noone knows how dundi is used here. weird |
| 23:29.45 | WIMPy | bio-tty: Or didn;t understand your question. |
| 23:30.19 | saxa | WIMPy: nothing happens |
| 23:30.30 | saxa | vmexten=900 ? |
| 23:30.38 | bio-tty | right |
| 23:30.38 | saxa | is that correct ? in sip.conf ? |
| 23:30.44 | WIMPy | yes |
| 23:30.47 | saxa | ok |
| 23:30.54 | saxa | still nothing then |
| 23:31.21 | saxa | any other ideas what can it be ? |
| 23:31.54 | leejohn | bio-tty, dundi is used for call routing without a centralized system like enum. |
| 23:31.56 | WIMPy | An evil phone. |
| 23:32.55 | leejohn | saxa, : could you pastebin the relevant section of the phone |
| 23:32.56 | bio-tty | leejohn: so then you route your call there and then what? |
| 23:33.00 | leejohn | +++++++++++++++++++++++9+++++++++++++++++++++ |
| 23:33.51 | WIMPy | bio-tty: I still don't know what your real question is, but dundi and enum are used to obtain routing information. |
| 23:33.52 | bio-tty | leejohn: you have to have a billing deal with the peer you route it to maybe. i start to get it |
| 23:34.00 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 23:34.21 | leejohn | bio-tty, : i will present you some scenario what dundi can be useful for your |
| 23:34.38 | bio-tty | leejohn: that would be great |
| 23:35.31 | leejohn | bio-tty, : imagine you have a couple of dozen of peers (sip trunk) then on administration point of view |
| 23:36.11 | leejohn | bio-tty, it's too hard to memorize those prefixes or locals on your network and you have no control on the other asterisk server |
| 23:36.45 | leejohn | bio-tty, the other administrator on the other end will publish his/her prefix in the dundi context |
| 23:37.26 | leejohn | bio-tty, that prefixes will automatically propagate on the dundi peers automatically |
| 23:38.05 | bio-tty | leejohn: this is in the public dundi is it? |
| 23:38.17 | leejohn | bio-tty, : not necessarily |
| 23:38.44 | leejohn | bio-tty, dundi is equivalent to e.num in asterisk world |
| 23:39.10 | bio-tty | leejohn: so my peer and me use our own dundi context, which floats around in the public dundi? |
| 23:39.30 | leejohn | bio-tty, yes |
| 23:39.45 | WIMPy | Probably not. |
| 23:39.54 | Qwell | there is no "public dundi" |
| 23:40.16 | WIMPy | thinks dundi is mostly used in a private set of peers. |
| 23:40.24 | leejohn | is stand corrected. it's not public |
| 23:40.33 | WIMPy | Well there was that public peering agreement. |
| 23:40.40 | Qwell | 8 years ago. |
| 23:40.48 | WIMPy | Or whatever iot was called. |
| 23:40.55 | bio-tty | i wondered about the idea of the public agreement, in respect to billing. |
| 23:41.25 | WIMPy | I think it's pretty dead. |
| 23:41.36 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:42.13 | bio-tty | oh, i wanted to find out about dundi, and i find this GBA(tm) everywhere, for example dundi.com |
| 23:42.24 | bio-tty | GPA(tm) |
| 23:42.46 | bio-tty | i didnt quite understand how billed targets would fit in. |
| 23:43.17 | WIMPy | What kind of billing are you talking about? |
| 23:43.20 | leejohn | bio-tty, : we have 6 dundi peers on our own network for ease of administration, but it's purely for call routing purposes, no billing or CDR involved |
| 23:43.32 | WIMPy | Premium rate services? |
| 23:44.56 | bio-tty | hmm. if i had an asterisk that gets incoming calls to be routed to my own set of pstn-gatewaying asterisk, maybe i start to see the use of dundi |
| 23:45.12 | bio-tty | s/incoming// |
| 23:45.34 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |