| 00:04.01 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 00:15.20 | *** join/#asterisk iCEBrkr (~icebrkr@cyberdyne.org) |
| 00:18.57 | pdtpatrick | Question - is it possible to have a clustered Asterisk environment. I would like a secondary so that in case primary goes down - the other picks up. But I have quemetrics running. How can both these system attach to the same quemetrics without destroying each other? |
| 00:24.27 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 00:29.52 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 00:31.09 | moniker326 | what cdr works with 1.8... I upgraded to 1.8 asterisk and now the cdr isn't recording calls again.. what do I need to do to fix this? |
| 00:32.00 | moniker326 | I looked for asterisk-addon-1.8 but no avail |
| 00:35.50 | WiretapWork | moniker326, asterisk packages are not supported, prefer compiling from source |
| 00:36.01 | WIMPy | Addos have been included wich you could have seen when doing make menuselect. |
| 00:36.39 | WIMPy | Did you read the upgrade* files? |
| 00:37.02 | WIMPy | And you should at least tell us how you try to store your cdrs. |
| 00:42.47 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 00:43.13 | *** part/#asterisk wwalker (~wwalker@208.92.232.27) |
| 00:48.24 | moniker326 | ok well yes I followed the instructions from the link n3 sent me last week but I don't have the exact one now |
| 00:48.40 | moniker326 | and I didn't notice the cdr wasn't recording until just today |
| 00:49.14 | moniker326 | wimpy: I Just use mysql to store the cdrs from the defaults |
| 00:49.18 | moniker326 | nothing special |
| 00:52.26 | sawgood | what is the purpose for extensions.ael in Asterisk 1.8? |
| 00:52.51 | *** join/#asterisk allan8904 (~allan@unaffiliated/allan8904) |
| 00:59.11 | *** join/#asterisk resno (~bryan@unaffiliated/resno) |
| 01:01.46 | *** join/#asterisk MDesade (~chatzilla@ip24-251-93-137.ph.ph.cox.net) |
| 01:01.52 | resno | is asterisk-gui still reliable? it seems not work for me. even though "make checkconfig" says its working fine |
| 01:02.18 | resno | ariel_: hey you here? |
| 01:07.31 | jeffspeff | i'm having the worst brain fart, i haven't done asterisk in years... when setting up my sip provider, the outgoing settings go into sip.conf, what about the inbound? |
| 01:10.09 | carrar | stinky |
| 01:10.34 | carrar | same file |
| 01:15.07 | *** join/#asterisk Kumbang (~unknown@180.245.137.5) |
| 01:25.57 | p3nguin | sip.conf controls all of SIP, regardless of which way a call is going. |
| 01:30.35 | moniker326 | what is needed to be known to make the cdr working in 1.8 from and upgrade from 1.6? |
| 01:31.39 | sawgood | -- Registered SIP '25079' at 215.43.75.189:5060 |
| 01:31.39 | sawgood | <PROTECTED> |
| 01:32.01 | sawgood | with only one truck setup on an IP PBX ... why would both these statements arrive at the ITSP side? |
| 01:32.52 | sawgood | they are arriving back to back about every 4 or 5 minutes to the ITSP registration server |
| 01:36.00 | WiretapWork | because you have two devices registering? |
| 01:37.43 | sawgood | only one trunk on the IP PBX? |
| 01:40.04 | WiretapWork | is your phone registering direct to the ITSP too? |
| 01:40.47 | sawgood | let me check |
| 01:41.57 | sawgood | maybe someone has a stand alone box or phone on the same LAN trying to register to the ITSP at the same time (I'll ask around) |
| 01:42.02 | sawgood | very strange |
| 01:43.48 | sawgood | I do not know how they could have / would have gotten the correct username and password, but I guess it is possible ... |
| 01:44.28 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 02:05.26 | *** join/#asterisk Scorp1us (~as@pool-72-81-130-240.bltmmd.fios.verizon.net) |
| 02:05.44 | Scorp1us | how in freepbx do I set up AGI destinations? |
| 02:12.23 | SunTsu | the way it's done in freepbx, while this is an asterisk channel |
| 02:16.18 | Scorp1us | ah |
| 02:19.21 | ariel_ | resno: I am here now. |
| 02:20.18 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 02:24.13 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:24.36 | DrDigital | i wonder why my irc client doesnt connect to this channel automaticly like the rest |
| 02:24.58 | DrDigital | i have it set to automaticly identify |
| 02:25.23 | DrDigital | seems like it identifies after it trys to join |
| 02:25.26 | DrDigital | #asterisk :Cannot join channel (+r) - you need to be identified with services |
| 02:25.41 | DrDigital | then right after, gallery/DrDigital :is now your hidden host (set by services.) |
| 02:27.03 | WiretapWork | DrDigital, set it up as server pass rather than nick pass |
| 02:27.48 | DrDigital | doubt that do anything? as its hard coded in x-chat aqua |
| 02:27.58 | DrDigital | its hard coded to message nickserv |
| 02:28.08 | DrDigital | ill try |
| 02:28.34 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:28.48 | DrDigital | wild |
| 02:29.41 | DrDigital | it worked |
| 02:30.10 | WiretapWork | DrDigital, see, tolja :P |
| 02:30.57 | WiretapWork | if you want it to auth on your alternate nicks, use the format ':accountname password' |
| 02:35.09 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 02:46.00 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 02:46.18 | ariel_ | hello, in asterisk 1.4.36 is there any reason sip would not even attempt to register with another server? Is there a way to force an attempt? |
| 02:49.45 | jeffspeff | i'm doing something stupid with my sip.conf file, but I just can't figure it out... any help would be great, i'm getting tired of staring at this thing... http://pastebin.com/cLKHeZ2X |
| 02:57.44 | *** join/#asterisk byronc (~byron@byron.theclarkfamily.name) |
| 02:59.15 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:04.44 | *** join/#asterisk nightrid3r (~kvirc@91.176.177.198) |
| 03:04.47 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 03:11.45 | jeffspeff | i'm doing something stupid with my sip.conf file, but I just can't figure it out... any help would be great, i'm getting tired of staring at this thing... http://pastebin.com/cLKHeZ2X |
| 03:17.21 | kaldemar | jeffspeff: start by changing your peer from type=user to type=peer. then please tell what the problem is. |
| 03:18.13 | *** join/#asterisk ajkaanbal (~ajkaanbal@189.181.28.21) |
| 03:18.35 | kaldemar | also remove type= friend from general. |
| 03:24.31 | jeffspeff | kaldemar, thanks, i made those changes, but i'm still getting the following error from sip debug --> Looking for s in demo (domain 173.11.144.149:15185) |
| 03:25.59 | *** join/#asterisk WiretapWork_ (~Wiretap@unaffiliated/wiretap) |
| 03:45.20 | *** join/#asterisk DrDigi (~mmurphy@c-67-181-109-249.hsd1.ca.comcast.net) |
| 03:51.34 | *** join/#asterisk luckyaba (~Lucky@ip72-194-218-169.sb.sd.cox.net) |
| 03:51.38 | kaldemar | jeffspeff: that's not an error. add /yournumber to the register statement and a context parameter for the peer. |
| 03:52.30 | jeffspeff | kaldemar, so the context setting goes to the peer not under general? |
| 03:55.08 | luckyaba | I am able to originate a call via .call file with Set:FOO=123 but when I attempt to originate a call with the same info via a script or directly through the AMI it comes back with "1 failed so falling back to exten 's'" |
| 03:55.14 | *** join/#asterisk sourcode (~code@ppp-58-8-96-210.revip2.asianet.co.th) |
| 03:55.23 | kaldemar | it can be under both, but you'd want to set a unique context for the peer. |
| 03:55.47 | luckyaba | its the same info in the .call file as directly through the AMI but for whatever reason the AMI directly or the script won't pass the variable through |
| 03:55.58 | jeffspeff | kaldemar, i'm still getting the same error, but this time it has my number instead of an s |
| 03:56.40 | kaldemar | jeffspeff: it's still not an error. make an extension that matches the number. |
| 03:57.17 | jeffspeff | isn't it supposed to default to s though? as in --> exten => s,1,Wait(1) |
| 03:57.30 | kaldemar | luckyaba: what else does your call file have? |
| 03:57.40 | kaldemar | jeffspeff: no. |
| 03:58.41 | kaldemar | jeffspeff: s is used when there is no incoming extension. you have an extension, i.e. your number. |
| 03:59.25 | jeffspeff | kaldemar, so in extensions.conf i just need to make a context to accept that call and then direct it where i need it to go? |
| 04:00.24 | luckyaba | kaldemar, Action:originate |
| 04:00.25 | luckyaba | Channel:SIP/voipms/5555555555 |
| 04:00.25 | luckyaba | Priority:1 |
| 04:00.25 | luckyaba | MaxRetries:2 |
| 04:00.25 | luckyaba | RetryTime:60 |
| 04:00.25 | luckyaba | WaitTime:30 |
| 04:00.29 | luckyaba | Context:conf |
| 04:00.32 | luckyaba | Extension:123 |
| 04:00.33 | luckyaba | Set:FOO=1234 |
| 04:01.04 | luckyaba | literally the only difference is obviously in the AMI I add the action:originate |
| 04:01.10 | kaldemar | ~pb |
| 04:01.11 | 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.us , 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. |
| 04:01.13 | luckyaba | yeah sorry |
| 04:01.29 | luckyaba | I got a bit excited that I such a quick response |
| 04:01.37 | luckyaba | i got ** |
| 04:02.22 | kaldemar | jeffspeff: yes. extensions.conf is the dialplan that lets asterisk know what to do with calls. |
| 04:03.04 | jeffspeff | i guess i'm not understanding why i have to specify my phone number as an extension on my register command |
| 04:05.35 | kaldemar | jeffspeff: so that you know what number the call comes in to. when you dialplan does more than Wait(1), it's valuable information. of course it is up to you to design your system. |
| 04:06.33 | kaldemar | luckyaba: pastebin your dialplan, the call file and what you see in CLI when you originate the call. |
| 04:07.35 | jeffspeff | kaldemar, my point is that it should still work without my number at the end by going to the s extension of the default context in extensions.conf right? all i get when i call my number is a busy signal |
| 04:14.13 | *** join/#asterisk radic (~radic@dslb-178-002-232-158.pools.arcor-ip.net) |
| 04:14.15 | luckyaba | kaldemar, http://pastebin.com/XYy1FLCW |
| 04:17.01 | WiretapWork_ | DrDigi, made it back I see, did the VPN work/ |
| 04:19.12 | *** join/#asterisk blake (~blake@2607:f8b8:2:3:129a:ddff:feb1:9b59) |
| 04:20.36 | luckyaba | kaldemar, Is there some additional arguments I should be sending via the script to specify where exactly in the dialplan to send the call? |
| 04:20.36 | DrDigi | nope |
| 04:20.44 | DrDigi | trying to get help in #pfsense |
| 04:21.52 | kaldemar | luckyaba: context, extension and priority are it. |
| 04:22.18 | luckyaba | thats what I thought... |
| 04:22.26 | kaldemar | jeffspeff: it should work if you have a matching extension in the context that the call lands in. |
| 04:22.54 | luckyaba | so why on earth would the .call file act differently than a script or a telnet session directly to the AMI |
| 04:22.54 | Nugget | telnet is eeeeeeevil! |
| 04:22.59 | luckyaba | hahaha |
| 04:23.51 | kaldemar | luckyaba: are you lacking the extension in the failed attempts? |
| 04:24.35 | jeffspeff | is this normal? From: <sip:starengr@den.teliax.net>;tag=as55b43965 |
| 04:24.35 | jeffspeff | <PROTECTED> |
| 04:25.01 | kaldemar | is what normal? |
| 04:25.19 | jeffspeff | the from and to having the same name |
| 04:25.27 | kaldemar | when? where? |
| 04:25.51 | jeffspeff | found it in my sip debug after i did a reload command |
| 04:26.23 | luckyaba | kaldemar, lmao.... ok so apprently the AMI doesn't recognize "extension" but it does "exten" |
| 04:26.31 | luckyaba | but it isn't taking the SET option now |
| 04:27.55 | kaldemar | luckyaba: there is no Set in manager originate. see "manager show command originate" in CLI. |
| 04:28.19 | kaldemar | "Variable" is what you want |
| 04:30.24 | luckyaba | lovely! |
| 04:35.26 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 04:37.55 | Scorp1us | i'm trying to debug my test.py agi and its not going well |
| 04:38.34 | Scorp1us | I'm using pyst and trying to use agi.say_date(time.time(), '') |
| 04:38.43 | Scorp1us | and all it does is hang up on me |
| 04:40.28 | irroot | scorp1us do me fav set verb 3 and paste the output had similar problem yesterday also please tell me what line types/codecs are in use |
| 04:42.22 | Scorp1us | <PROTECTED> |
| 04:42.22 | Scorp1us | <PROTECTED> |
| 04:42.30 | Scorp1us | that's all there is |
| 04:42.47 | Scorp1us | even with agi set debug on |
| 04:43.06 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-bavaxmtaliaptymt) |
| 04:43.18 | irroot | and the script is running as expected no format errors or other problems |
| 04:43.39 | Scorp1us | well I am not sure. I'm trying to debug that |
| 04:43.54 | Scorp1us | nothing is obvious though |
| 04:44.04 | irroot | problem i was having was the channel would hang up incompat codecs |
| 04:44.10 | Scorp1us | ah |
| 04:45.10 | Scorp1us | how do i check that out? |
| 04:45.46 | irroot | it does not look like it applies as in the console it would warn b4 disconnect |
| 04:45.55 | irroot | looks like you got a "dud" script |
| 04:46.12 | Scorp1us | i wrote it |
| 04:46.19 | Scorp1us | why is it a dud? |
| 04:46.30 | irroot | its not running |
| 04:46.42 | Scorp1us | it returns zero |
| 04:46.45 | irroot | put some quotes in there |
| 04:46.53 | irroot | trace through it |
| 04:47.18 | irroot | prolly a logic / conditional statment that is wrong / ignored |
| 04:49.09 | Scorp1us | its super short: http://pastebin.com/Ma4FVQJX |
| 04:52.42 | luckyaba | kaldemar, seriously man, I appreciate the help |
| 04:52.49 | luckyaba | got it all working now |
| 04:54.20 | kaldemar | luckyaba: no problem. |
| 04:58.43 | Scorp1us | wahoo! i got it working |
| 04:58.47 | Scorp1us | thanks everyone |
| 04:58.49 | irroot | cool |
| 04:58.58 | Scorp1us | it was a bum script. |
| 04:59.21 | Scorp1us | bed time |
| 04:59.38 | *** join/#asterisk hrolf (~hrolf@202.61.49.9) |
| 04:59.39 | irroot | :P its 06:59 here |
| 04:59.49 | irroot | in for the day |
| 05:02.31 | sawgood | Is there a way, I could make this dialplan statement work for all contexts without manually entering it in everyone? |
| 05:02.35 | sawgood | exten => h,1,NoOp(Hangupcause${HANGUPCAUSE}) |
| 05:02.59 | irroot | use include ?? |
| 05:03.06 | sawgood | thank you |
| 05:03.13 | irroot | have a new context with just the h bits |
| 05:03.20 | irroot | and do a global h exten |
| 05:03.37 | irroot | include => my_hangup_ctx |
| 05:03.45 | sawgood | what is a global h exten? |
| 05:04.17 | irroot | you will effectivly have onw |
| 05:04.22 | irroot | one |
| 05:05.13 | sawgood | ty |
| 05:09.23 | sawgood | wow that worked ... thank you very much |
| 05:11.07 | sawgood | cool now I have all calls logging what the hangupcause code is ... |
| 05:14.12 | irroot | pleasure |
| 05:17.47 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
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| 05:55.15 | hrolf | In Asterisk, if STREAM FILE cannot find a given file, then it returns '200 result=0 endpos=0', should I consider it as error? |
| 05:55.59 | *** join/#asterisk jetlag (~jetlag@pool-71-168-244-171.cmdnnj.east.verizon.net) |
| 05:56.41 | hrolf | It returns '200 result=0 endpos=0' to the AGI client, but there is no way to know if the file hasn't been played, so my question, should I take the endpos=0 as a error? |
| 06:04.55 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 06:13.00 | nightrid3r | what gets returned when all goes fine? |
| 06:13.32 | *** join/#asterisk Ad-Hoc (~nimbus@62.169.216.185) |
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| 06:13.48 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 06:19.29 | hrolf | nightrid3r: the same thing except that the endpos contains the frame where the playback stopped |
| 06:19.44 | hrolf | nightrid3r: mostly, not endpos=0 |
| 06:19.55 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:20.03 | nightrid3r | then you can indeed use the endpos as an error |
| 06:20.23 | hrolf | nightrid3r: but I have one concern, that if STREAM FILE is stopped by pressing a DTMF then endpos=0 can possibly be 0 |
| 06:20.35 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:20.57 | nightrid3r | true but how big is that chance? |
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| 06:41.41 | Russ | I'm looking to update from 1.6.2 to 1.8, but I'm using asterisk-flite |
| 06:41.59 | Russ | and I don't see debian packages for asterisk-flite that work with 1.8 |
| 06:42.05 | Russ | is anyone using asterisk-flite with 1.8? |
| 06:43.47 | *** join/#asterisk coppice (~coppice@m121-202-29-99.smartone-vodafone.com) |
| 06:50.21 | kaldemar | Russ: it works with 1.8 just fine. compile it yourself if you can't find a package. |
| 06:50.37 | Russ | thanks |
| 06:52.29 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:52.39 | schmidts | good morning |
| 06:52.58 | Russ | hmmm...I'll probably need to recompile my g72x modules |
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| 07:17.51 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 07:18.34 | hrolf | Is there any input command which takes escape digits (after pressing which, the input is terminated) and a timeout parameter of unlimited size? |
| 07:18.52 | Russ | bah, dahdi failed to build against 2.6.39 |
| 07:21.52 | hrolf | there is Get Option but it returns only one character (I donot mention any file name, so it works for input), there is Get Data but it doesn't terminate on any specific character (like if I want it to stop after # is pressed) |
| 07:25.02 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:27.58 | sezuan | Can someone point to a resource which explain the difference between chan_gtalk and chan_jingle? I thought google uses (more or less) jingle, but after reading ASTERISK-17993 it seems that gooletalk is more like a proprietary protocol. |
| 07:27.59 | CaptainPants | [ASTERISK-17993] [Status: Open] Inbound Google Chat calls fail - https://issues.asterisk.org/jira/browse/ASTERISK-17993 |
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| 07:58.25 | Russ | ah, there is a new package, asterisk-dahdi |
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| 07:59.02 | madmax78 | has anybody used xorcom asterisk patch in here |
| 08:00.02 | Russ | ah, asterisk 1.6-1.8 and linux 2.6.32->2.6.39 upgrade complete :) |
| 08:00.41 | irroot | is trying to avoid road trip .... who goes to v-town any way |
| 08:01.09 | Russ | "v-town"? |
| 08:01.16 | Russ | is that like "toon-town"? |
| 08:04.53 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
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| 08:06.09 | joako | Anyone knows why GotoIf doesn't work as documented in Asterisk 1.6? |
| 08:06.28 | ChannelZ | how so |
| 08:06.43 | ChannelZ | works here |
| 08:07.45 | joako | I follow the documentation and the dialplan continues as if the command wasn't there: GotoIf("SIP/channel-0000003a", "SUCCESS = SUCCESS?macro-vmcenter,s,100") |
| 08:07.55 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:f944:7cdf:ec61:2f22) |
| 08:08.39 | ChannelZ | you probably have a syntax problem |
| 08:10.29 | ChannelZ | Asterisk expressions get real ugly real fast, they're easy to screw up |
| 08:10.47 | joako | Do you know where I can find the proper syntax? GotoIf(condition?[labeliftrue][:labeliffalse]). I'm not sure how I could mess that up. |
| 08:11.07 | ChannelZ | That's the correct syntax, but YOUR syntax is probably wrong |
| 08:11.34 | ChannelZ | your comparison (the 'condition' part) I'm sure |
| 08:12.12 | joako | exten => s,7,GotoIf(${VMBOXEXISTSSTATUS} = SUCCESS?macro-vmcenter,s,100). Executed in the console is: GotoIf("SIP/channel-0000003a", "SUCCESS = SUCCESS?macro-vmcenter,s,100") |
| 08:12.32 | ChannelZ | yeah. |
| 08:12.52 | ChannelZ | $[${VMBOXEXISTSSTATUS}=SUCCESS] |
| 08:13.37 | ChannelZ | notice it's wrapped in $[ ] |
| 08:14.09 | joako | Yes. I started off with something from the crappy voip-info wiki |
| 08:14.37 | ChannelZ | exten => s,7,GotoIf($[${VMBOXEXISTSSTATUS}=SUCCESS]?macro-vmcenter,s,100) |
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| 08:33.09 | joako | ChannelZ, It's working. Thanks for the help. |
| 08:37.02 | *** join/#asterisk Cadey (~x@62.84.178.106) |
| 08:37.32 | *** join/#asterisk engrxyz (~puitpyitr@212.23.51.7) |
| 08:38.29 | Cadey | Hi guys, not sure if you can help or not as its my suppision its a Aastra thing and not an asterisk thing bt here does. When my users are dialing incomplete numbers our provider returns a 484 "Address Incomplete" however the phone renders this as a busy tone when it should really in my opinion state its an unavaliable number. Any idears? |
| 08:38.44 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 08:39.32 | joako | Cadey, Disable early dial support in the phones. |
| 08:40.45 | Cadey | so kind of like putting dial patters on the phones to stop that making invalid calls |
| 08:40.50 | Cadey | and not rely on the provider? |
| 08:41.21 | irroot | does not like yealink phones GRRRRR |
| 08:41.42 | joako | Cadey, Correct. You don't even *need* a custom dialplan. The default should work, but you might have to either press send or wait a few seconds after dialing before the call connects. |
| 08:41.44 | Cadey | has 3 yealink phones for emergency calls |
| 08:42.06 | Cadey | joako, the default is x+#|xx+* :) |
| 08:42.12 | Cadey | which is more a less anything lol |
| 08:42.36 | joako | If you get the dial plan working it's nice, e.g. on my polycom here I can dial *67 (block caller id), get a 2nd/stuttered dialtone, dial 10 digit number and it starts to connect instantly |
| 08:47.15 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 09:13.37 | *** join/#asterisk NourSs (~gholzinge@LAubervilliers-151-13-22-64.w217-128.abo.wanadoo.fr) |
| 09:13.42 | *** part/#asterisk NourSs (~gholzinge@LAubervilliers-151-13-22-64.w217-128.abo.wanadoo.fr) |
| 09:18.16 | madmax78 | question: i have rtp ports 11000 - 20000 set in rtp.conf and port 5060 for sip. i have peers connected to port like 65117, 49430 an such, how is that possible |
| 09:19.07 | kaldemar | madmax78: what makes you think they connect TO port 65117 or 49430? |
| 09:19.21 | madmax78 | sip show peers say so |
| 09:20.09 | kaldemar | madmax78: it says where they connect FROM. |
| 09:20.21 | madmax78 | alright |
| 09:20.30 | madmax78 | thanx |
| 09:24.04 | madmax78 | so the FROM port still connects to 5060? |
| 09:28.16 | madmax78 | thew sip show peer port is showing the rtp port, right? |
| 09:30.23 | *** join/#asterisk TimeRider (~steve@host-92-27-131-175.static.as13285.net) |
| 09:31.29 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:33.43 | kaldemar | madmax78: they connect to whatever your asterisk listens to. in this case, 5060. and sip show peers shows a SIP port, not RTP. |
| 09:36.13 | *** join/#asterisk coppice (~coppice@m121-202-29-99.smartone-vodafone.com) |
| 09:39.43 | *** join/#asterisk lampe (5264d142@gateway/web/freenode/ip.82.100.209.66) |
| 09:39.49 | lampe | hi |
| 09:40.35 | lampe | I am using a starface.. and I lost my admin password for the webinterface.. I guess the procedure for restoring it is similar to asterisk.. any hints? |
| 09:41.55 | Wiretap7 | asterisk doesn't have a web interface |
| 09:42.42 | joako | Wiretap7, It does. But it uses the manager credentials. |
| 09:46.22 | kaldemar | asterisk doesn't have a web GUI. what is starface using? |
| 09:47.57 | cneb3000 | lampe: have a look in /etc/asterisk/manager.conf and see if those credentials work. |
| 09:50.12 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:50.25 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:52.39 | lampe | cneb3000 no doesnt work |
| 09:55.12 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 09:56.14 | cneb3000 | lampe: there should be a few blocks of text like [admin] or something. the word inside the [ ]s is the username (so [admin] = admin) and the password is the 'secret'. |
| 09:56.16 | schmidts | does anybody know if there is a sip client for a smartphone (apple or android) which supports G729? |
| 09:56.25 | cneb3000 | lampe: with that in mind, are you sure it doesn't work? :) |
| 10:01.28 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 10:09.30 | lampe | cneb3000 no it really doesnt work.. |
| 10:10.43 | kaldemar | lampe: find out what GUI it uses, then you know the right place to look for and the right place to ask. |
| 10:12.37 | *** join/#asterisk Gugge (~gugge@91.208.16.1) |
| 10:29.02 | lampe | it seems to be a tomcat without an apache.. |
| 10:33.18 | sgimeno | schmidts: http://www.counterpath.com/bria-android-edition.html ? |
| 10:47.32 | kaldemar | lampe: no reason to expect that it has something in common with asterisk then. |
| 10:47.48 | *** join/#asterisk CoderForLife (~Miranda@unaffiliated/coderforlife) |
| 10:48.49 | lampe | kaldemar for the GUI frontend.. |
| 10:51.08 | *** join/#asterisk saeidw (~user@46.227.248.114) |
| 10:54.29 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 10:59.41 | lampe | I guess it is stored in an own db..thx for commenting my search :) |
| 11:00.53 | saeidw | Hello everyone, I'm having a bit of trouble with chan_mobile, and I just want to make sure that it's not because I missed a fundamental concept |
| 11:02.01 | saeidw | I want asterisk to answer calls on my mobile phone and playback a message |
| 11:06.25 | saeidw | I set up my chan_mobile.conf and extensions.conf and was able to pair with a phone. |
| 11:07.08 | saeidw | but when I call the phone, it immediately hangs up, and the debug messages show a failure |
| 11:07.34 | *** join/#asterisk whirlpbx_ (~whirlpbx@212.58.196.250) |
| 11:13.14 | WIMPy | Nothing more than "failure"? |
| 11:13.27 | saeidw | Let me copy the message |
| 11:13.37 | *** join/#asterisk florin_m (bc2721b6@gateway/web/freenode/ip.188.39.33.182) |
| 11:13.40 | cneb3000 | saeidw: paste it on pastebin.com :) |
| 11:13.42 | florin_m | hi there |
| 11:13.45 | saeidw | ok |
| 11:13.47 | florin_m | got a small problem |
| 11:13.54 | cneb3000 | thats what she said |
| 11:14.07 | florin_m | how to extract 'm=video 24002 RTP/AVP 34' from a sip header ? |
| 11:14.48 | florin_m | i need to check if a sip call is a video call or audio call |
| 11:15.42 | cneb3000 | florin_m: has the call already happened? can you re-create it? |
| 11:15.47 | florin_m | and seems that if a sip call is a video call then : Peer video RTP is at port ... and if is not a video call : Peer doesn't provide video |
| 11:15.56 | florin_m | not already happened |
| 11:16.20 | florin_m | in the extention.conf as is initiated i want to do different things if is a video or audio only call |
| 11:16.26 | cneb3000 | if you're just trying to read sip headers, I personally use a wireshark trace to do that. |
| 11:16.59 | cneb3000 | hm, oh you mean.. like the dial plan detecs if it's video or not and does something because of that? |
| 11:17.06 | florin_m | yep |
| 11:17.18 | florin_m | i noticed that if ${SIP_HEADER(Content-Length)}) > 500 is a video call |
| 11:17.28 | florin_m | if is 3xx then is audio call |
| 11:17.33 | florin_m | but i don't like this :) |
| 11:17.39 | florin_m | need to be more preceise |
| 11:18.00 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
| 11:18.28 | florin_m | so any idea ? |
| 11:18.45 | *** join/#asterisk radic (~radic@dslb-178-002-232-158.pools.arcor-ip.net) |
| 11:18.53 | cneb3000 | hm, not from me. I was just reading the documentation |
| 11:19.39 | saeidw | ok, here's a paste of chan_mobile failure: |
| 11:19.44 | florin_m | so i see headers of the sip |
| 11:19.55 | florin_m | how to acces the content of the sip? |
| 11:20.09 | saeidw | http://pastebin.com/NrX7jsRB |
| 11:20.35 | florin_m | http://pastebin.com/4id4PcsH this is a video call |
| 11:20.58 | WIMPy | florin_m: function SIP_HEADER |
| 11:21.07 | kaldemar | florin_m: m=video 24002 RTP/AVP 34 is in the SDP part, it's not in the SIP headers. |
| 11:21.38 | WIMPy | saeidw: That's not a failure. You just don't have an s extension. |
| 11:21.48 | florin_m | so how so how can i check if is a video or not? any ideas? |
| 11:22.02 | WIMPy | Hmm. Missing SIP_BODY? |
| 11:23.58 | florin_m | on a isdn is simple : $[${CHANNEL(transfercapability)}=VIDEO]?video:digital) |
| 11:24.09 | florin_m | but on sip i didn't find any kind of info :( |
| 11:24.31 | WIMPy | Use H.323 instead of SIP. |
| 11:24.43 | florin_m | i can't |
| 11:24.50 | florin_m | my handset support only sip |
| 11:26.50 | florin_m | ok found a nastrier trick :) |
| 11:26.58 | c0rnoTa | florin_m: what about looking for supported codecs on device? |
| 11:27.03 | florin_m | ${CHANNEL(videonativeformat)} |
| 11:27.33 | c0rnoTa | yeap, like this |
| 11:27.39 | florin_m | the device can make audio and video calls |
| 11:27.44 | florin_m | if he initiate a video call |
| 11:27.58 | florin_m | then videonativeformat will will h263 or h264 |
| 11:28.01 | kaldemar | florin_m: check if there is ${CHANNEL(rtpdest,video)} |
| 11:29.14 | florin_m | is no video will answer 0.0.0.0 |
| 11:29.24 | florin_m | 0.0.0.0:0 |
| 11:29.28 | florin_m | in fact |
| 11:29.37 | florin_m | good shout |
| 11:29.42 | florin_m | i can use this ;) |
| 11:29.44 | Cadey | any people used to UK dial plans here? |
| 11:29.51 | *** join/#asterisk saeidw (~user@46.227.248.114) |
| 11:29.56 | florin_m | let me check some new other check |
| 11:30.18 | saeidw | sorry about that, I got disconnected |
| 11:31.45 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 11:32.29 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 11:32.45 | florin_m | got one more sick problem now :D |
| 11:32.53 | florin_m | voxbone |
| 11:32.58 | florin_m | if i 3g video call it |
| 11:33.23 | florin_m | it sends the 3g stream as a alaw stream |
| 11:33.31 | florin_m | all encapulated there like a isdn stream |
| 11:33.36 | WIMPy | saeidw: That's not a failure. You just don't have an s extension. |
| 11:34.14 | saeidw | I can paste in my extensions.conf if that helps |
| 11:34.32 | saeidw | I didn't see anything about an s extension in the documentation |
| 11:34.43 | WIMPy | saeidw: No. Just create an s extension. |
| 11:35.10 | WIMPy | saeidw: If you don't get a called party number, the call goes to s. |
| 11:35.13 | kaldemar | saeidw: then the documentation you used lacked the dialplan part. |
| 11:36.41 | WIMPy | As the mobile doesn'e even know its own number, it cant send it to Asterisk. |
| 11:37.48 | saeidw | oh, I just found it on the wiki under Auto Attendant and IVR Menus -> Handling Special Extensions |
| 11:37.57 | saeidw | thanks for that, I didn't know this existed! :D |
| 11:38.01 | saeidw | I'll try it out now |
| 11:49.51 | Kobaz | oh man |
| 11:50.14 | Kobaz | i have asterisk do a nightly restart when convenient |
| 11:50.17 | Kobaz | i need to turn that off |
| 11:50.35 | Kobaz | it's been deadlocking on shutdown, and now i can't kill the process, had to reboot |
| 11:51.05 | cneb3000 | why dont you do a forced restart. i dont think a deadlock would get caught on that? |
| 11:51.06 | WIMPy | has deleted that cronjob 26 days ago. |
| 11:51.13 | irroot | ok so the yealink is working TLS / SRTP / HD |
| 11:51.21 | irroot | but dont like it much |
| 11:51.37 | WIMPy | But it's quite normal that Asterisk needs a -9. |
| 11:52.06 | Kobaz | kill -9 19529; kill -9 19529; kill -9 19529; kill -9 19529; |
| 11:52.56 | Kobaz | /etc/init.d/asterisk restart definitly wouldn't deadlock on shutdown |
| 11:53.03 | Kobaz | now i just have to hope the server boots up |
| 11:53.24 | WIMPy | restart on shutdown? |
| 11:54.10 | Kobaz | yeah it's probably locked up on the reboot |
| 11:55.46 | WIMPy | On shutdown I do a 'core stop gracefully' and then wait until it exits or a minute has passed. |
| 11:56.12 | Kobaz | yeah but that's the same thing isn't it? |
| 11:56.16 | Kobaz | it does all the module unloads |
| 11:56.23 | Kobaz | that's what i need to avoid |
| 11:56.34 | Kobaz | just kill it, to avoid shutdown deadlocks |
| 11:57.24 | WIMPy | After 60s it gets a 'core stop now' and 4s later a kill -9. The later hasn't faild so far. |
| 11:57.37 | Kobaz | yeah |
| 11:57.43 | WIMPy | You should find out what module is causing that. |
| 11:57.44 | Kobaz | that wouldn't have solved this problem |
| 11:57.54 | Kobaz | for whatever reason the asterisk process isn't killable |
| 12:00.07 | Kobaz | wasn't.... it's been power cycled now |
| 12:01.47 | Kobaz | kjhkjhjkhkjhkhj |
| 12:04.29 | russellb | those are letters |
| 12:04.49 | Kobaz | yeap |
| 12:04.56 | Kobaz | server's not booting |
| 12:05.11 | Kobaz | damn, i don't want to drive 2.5 hours |
| 12:06.38 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:08.57 | Kobaz | WIMPy: iax2 i think |
| 12:09.17 | Kobaz | i couldn't get any info this time though, since asterisk was totally dead and unkillable |
| 12:12.08 | Kobaz | last time it deadlocked it was iax2 though |
| 12:13.03 | WIMPy | I only had that with chan_lcr so far. On unload it never got a chance to kill itself. |
| 12:13.44 | Kobaz | oh sweet |
| 12:13.46 | Kobaz | server is up |
| 12:13.47 | Kobaz | whew |
| 12:13.59 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 12:19.03 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
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| 12:29.55 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-esgbuwgbklxessmo) |
| 12:34.32 | *** join/#asterisk saeidw (~user@46.227.248.114) |
| 12:38.08 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:40.56 | *** join/#asterisk wonderworld (~ww@port-92-201-73-185.dynamic.qsc.de) |
| 12:44.28 | *** join/#asterisk timholum (~tholum@75-121-149-228.dyn.centurytel.net) |
| 12:51.12 | *** join/#asterisk Sheeplet (~multivac@41-135-86-171.dsl.mweb.co.za) |
| 12:51.22 | Sheeplet | lo all |
| 12:55.39 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 12:58.30 | *** join/#asterisk coppice (~coppice@m121-202-29-99.smartone-vodafone.com) |
| 13:02.08 | irroot | F@#$%@#%@#$% MORONS .... |
| 13:02.10 | irroot | ok back to scheduled programing |
| 13:02.12 | irroot | some fool asks me to set up a IAX trunk to a telco but does not use trunking .... |
| 13:02.15 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-joewccpcjpnieeia) |
| 13:03.16 | cneb3000 | THAT'S NOT THEIR PROBLEM. JUST DO IT!! |
| 13:03.27 | cneb3000 | OR I'LL FIND SOMEONE ELSE THAT WILL |
| 13:03.28 | cneb3000 | :) |
| 13:05.06 | Katty | hi |
| 13:05.55 | Katty | eyes cneb3000 suspiciously |
| 13:07.17 | cneb3000 | Katty I have a present for you |
| 13:07.20 | cneb3000 | ~cheese |
| 13:07.20 | infobot | cheese is probably pure awesomesause. I love cheese. |
| 13:07.34 | WIMPy | LOL |
| 13:07.41 | Katty | i don't accept cheese from strangers. |
| 13:08.00 | WIMPy | I thin cneb3000 wanted to do |
| 13:08.05 | WIMPy | ~cheese Katty |
| 13:08.05 | infobot | ACTION hands Katty a tray of sliced sharp cheddar and pineapple |
| 13:08.17 | cneb3000 | i know what i meant to do ¬_¬ |
| 13:08.27 | Katty | hugs WIMPy |
| 13:08.29 | cneb3000 | in, "IRC fail" |
| 13:08.50 | WIMPy | What kind of cheesy present is it that you don't hand over? |
| 13:08.56 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 13:09.32 | Katty | i have a sausage biscuit anyway |
| 13:09.40 | Katty | it's vera nomable |
| 13:09.50 | cneb3000 | infobot, ATTATCK WIMPy FOR HIS PESTILENCE |
| 13:09.56 | irroot | ~beer |
| 13:09.57 | infobot | ACTION has disconnected (Read error: 99 (Connection reset by beer)) |
| 13:10.12 | WIMPy | No lets see... |
| 13:10.19 | WIMPy | ~beer cneb3000 |
| 13:10.20 | infobot | ACTION deftly decants a fine La Maudite for cneb3000 |
| 13:10.32 | Kobaz | well i figured out what totally destroys asterisk and makes it unkillable |
| 13:10.41 | Kobaz | lots of local channels in infinite loops |
| 13:10.46 | irroot | kobaz my techies ?? |
| 13:10.48 | Katty | hi Kobaz |
| 13:10.51 | Kobaz | hi Katty |
| 13:10.58 | WIMPy | I think infobot got his procedures mixed up. |
| 13:11.04 | cneb3000 | hehe |
| 13:11.33 | irroot | Katty why a biscuit ?? for me a biscuit is a "cookie" and a cookie oh well this is not the XXX chan |
| 13:12.15 | Katty | well it's an english muffin, really |
| 13:12.32 | Katty | with a sausage. |
| 13:12.43 | Katty | and we don't call cookies biscuits here |
| 13:12.50 | irroot | not a cookie with a sausage .... |
| 13:12.54 | Katty | no |
| 13:13.10 | Katty | i would probably die from sugar overload if it was a cookie with sausage for breakfast |
| 13:14.12 | Katty | (english muffin)++ |
| 13:14.25 | Kobaz | sugar |
| 13:14.31 | Kobaz | evil... beware |
| 13:14.52 | Kobaz | http://www.youtube.com/watch?v=dBnniua6-oM |
| 13:15.00 | Kobaz | not all sugars are created equal |
| 13:15.33 | irroot | is with kobaz eat more fat |
| 13:16.34 | *** join/#asterisk n0tk (~n0tk@gateway/tor-sasl/n0tk) |
| 13:16.57 | Kobaz | fat, and veggies |
| 13:17.05 | Kobaz | less sugar and carbs |
| 13:17.41 | irroot | lchf FTW |
| 13:22.53 | Katty | carbs aren't bad |
| 13:22.56 | Katty | it's the sugar that's bad |
| 13:23.10 | Katty | and sugar isn't really all that bad either, it's just the quantity we consume of it |
| 13:23.30 | Katty | sugar is bad tho. |
| 13:23.34 | irroot | Katty that is partially true carbs that are not processed or natural as in veggies |
| 13:23.35 | Katty | your liver will tell you so. |
| 13:23.51 | Katty | brown rice, nomnom |
| 13:23.55 | Katty | whole wheat pasta,nomnomnom |
| 13:24.04 | Katty | potato, nomnom |
| 13:24.14 | Katty | damnit i'm still hungry |
| 13:24.19 | Katty | curses metabolism |
| 13:24.34 | beek | Quinoa -- the ultimate grain. |
| 13:24.40 | *** join/#asterisk coppice (~chatzilla@210.17.219.183) |
| 13:24.52 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 13:24.53 | Katty | never had quinoa |
| 13:25.08 | beek | Oh Katty -- you've been missing out! |
| 13:25.09 | Katty | ever had ezekiel bread? |
| 13:25.24 | beek | nope |
| 13:25.49 | Katty | it's flourless |
| 13:25.56 | Katty | made completely from sprouted whole grains |
| 13:26.07 | Katty | and deeeelish |
| 13:26.14 | irroot | Katty beer bread with some swazi gold ?? |
| 13:27.35 | beek | So what did I miss installing with 1.8 to garner this: http://pastebin.com/XEBcTzDv |
| 13:28.37 | *** join/#asterisk scalex000 (~chatzilla@186.6.79.141) |
| 13:28.59 | Katty | ugg maybe sausage was a bad choice. |
| 13:29.02 | Katty | nauseous |
| 13:29.31 | *** part/#asterisk resno (~bryan@unaffiliated/resno) |
| 13:30.45 | bobb_WU | hey all- i have a network with multiple asterisk nodes and we have one set up as a relay. the nodes look locally and then forward to the relay if they don't find the extension. it the relay sees the number in a global view, it forwards to that box. at this point, is the relay node passing off the connection? how do i figure that out |
| 13:31.18 | bobb_WU | someone suggested that i look into canreinvite about this |
| 13:32.03 | Katty | hi there |
| 13:34.01 | bobb_WU | hi :) |
| 13:35.19 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:35.47 | Katty | hi jaytee |
| 13:37.45 | *** join/#asterisk bchia (~Adium@nat/digium/x-mmitjppgudvrjtjv) |
| 13:37.49 | Kobaz | Katty: how can it be flourless but made from whole grains |
| 13:37.58 | Kobaz | don't you make bread from some sort of flower? |
| 13:38.30 | coppice | they're grains of sand |
| 13:38.33 | Katty | http://www.foodforlife.com/our-products/sprouted-grain/ezekiel-49 |
| 13:39.03 | *** join/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net) |
| 13:39.24 | *** join/#asterisk Aas (~aas@static-dsl-88.87-197-156.telecom.sk) |
| 13:39.53 | MrTelephone | I'm using asterisk 1.8.4.2 and still have an ODBC voicemail renumbering issue (msgnum). Anyone else with the same problem? |
| 13:40.19 | pabelanger | MrTelephone: I believe that will be fixed in 1.8.5 |
| 13:40.24 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 13:40.27 | MrTelephone | ok |
| 13:41.09 | MrTelephone | It's not bad. I was going to write a perl script to fix on a cron job but it doesn't happen too often. Just happens to people who live on the phone |
| 13:41.43 | MrTelephone | 1.8.5 is probably due for release shortly? |
| 13:42.03 | *** join/#asterisk digiv_ (~mlhess@141.214.234.28) |
| 13:42.13 | pabelanger | MrTelephone: I believe 1.8.5-rc1 will be created to day |
| 13:42.20 | pabelanger | leifmadsen: ^ |
| 13:42.31 | leifmadsen | pabelanger: that's the plan |
| 13:42.37 | pabelanger | what what! |
| 13:42.41 | leifmadsen | ... |
| 13:42.42 | leifmadsen | you know what |
| 13:42.49 | *** join/#asterisk methodvon (~methodvon@pool-71-191-175-251.washdc.fios.verizon.net) |
| 13:43.09 | MrTelephone | I got an email about astricon. Is it a big sales pitch or do people share ideas there on how to make asterisk a better product? |
| 13:43.27 | leifmadsen | yes |
| 13:43.30 | *** part/#asterisk Aas (~aas@static-dsl-88.87-197-156.telecom.sk) |
| 13:43.46 | beek | MrTelephone: Been there last two years -- it's both. |
| 13:43.47 | leifmadsen | astricon is one of the few conferences I attend *because* it's not a big sales pitch |
| 13:44.03 | leifmadsen | of course there is commercial stuff going on, but that's the point of conferences |
| 13:44.14 | MrTelephone | Is it 695 for three days? the email wasn't really clear. the picing showed 495/595/695 or something like that. |
| 13:44.14 | pabelanger | and because Qwell buys the beer |
| 13:44.15 | beek | The sales pitch occurs in the commercial hall, but the sessions are techie. |
| 13:44.30 | leifmadsen | if you're there to see presentations though, there are some very high quality technical talks -- my talks never have sales things in them. Mine are all technical. |
| 13:44.36 | beek | Meeting with everyone is the high point, though. |
| 13:44.43 | leifmadsen | +1\ |
| 13:45.04 | bobb_WU | can somone help me out with passing calls to peers? |
| 13:45.10 | bobb_WU | i didn't get a response earlier |
| 13:45.17 | cneb3000 | ^^^ mrtelephony: if I can get the money together I'm thinking of going myself. look at some of the videos from previous years. |
| 13:45.47 | beek | So what have I missed specifying during compilating that has resulted in no documentation? http://pastebin.com/XEBcTzDv |
| 13:45.52 | beek | Version 1.8. |
| 13:46.20 | beek | libxml2-devel is there. |
| 13:46.21 | pabelanger | cneb3000: It is a good time |
| 13:46.27 | WIMPy | beek: Did you upgrade from an substantially older version? |
| 13:46.35 | beek | WIMPy: no... clean install. |
| 13:47.00 | beek | Just went through make menuselect... didn't see anything to specify that. |
| 13:47.03 | cneb3000 | pabelanger: expensive though :) not the tickets, it's everything else. |
| 13:47.11 | cneb3000 | (im from the UK) |
| 13:47.20 | beek | cneb3000: I regret that I have but one kidney to sell... |
| 13:47.28 | MrTelephone | I don't get to hang out with many nerds where I live so the conference should be worth it. |
| 13:47.29 | scalex000 | good morning, I need help to understand how to use codecs |
| 13:47.30 | cneb3000 | heh |
| 13:47.55 | scalex000 | I have ipphone using SIP and I connect 2 pbx with iax |
| 13:48.36 | scalex000 | to use codec g729 i need to setup on sip and iax |