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00:06.47 | atan2 | Guys having an issue calling voicemail using a grandstream adapter. I get "vm-password (format 0x1 (g723))" error message in console |
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00:08.45 | atan2 | I assuem the codec in the device is set wrong. Err, but not sure which to use. |
00:12.11 | atan2 | Oh great now the phones stream when I pick them up |
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00:19.50 | atan2 | err, my bad. Client left phone off hook. Figures. |
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00:34.17 | *** join/#asterisk GTXComm (~John@cpe-72-128-62-30.kc.res.rr.com) |
00:39.38 | GTXComm | Q I am running asterisk 1.6.0.6 as a provider for a Elastix system... Sip trunk setup, when I dial a to the other box it gives me a message chan_sip.c:15139 handle_response_invite: Failed to authenticate on INVITE to '"WIRELESS CALLER" <sip:8165298686@63.76.55.135>;tag=as64fd44e6'... I can call outbound, but it doesn't seem to want to take incoming registration on SIP. I use IAX2 on inbound, and it works, but not on SP inbound. The Ela |
00:39.38 | GTXComm | stix (FreePBx0 is version 1.4.19. I am not used to setting up asterisk to asterisk SIP. |
00:40.56 | GTXComm | 1st question. on this.. hope i am okay asking. |
00:41.25 | GTXComm | <--- new |
00:41.31 | GTXComm | here |
00:42.29 | GTXComm | Do I need to have a special registar string on both servers, on my sip.conf |
00:42.35 | GTXComm | ? |
00:46.30 | JerJer | you need like defaultuser= on the peer |
00:50.10 | GTXComm | Ok, on the other side do I need to have a recognizable DID on ibound? Because my sip is [c20024-101] so set up defaultuser=[c20024=101] ? |
00:50.49 | GTXComm | or sorry defaultuser=c20024-101 |
00:52.20 | GTXComm | That was confusing, let me start over .. On the 1.4 server, do I need defaltuser= , and 1.6 pot in Defaultuser=? |
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02:08.12 | RichardLynch | How can I use Dial macro with SIP and force it to go over TCP, without setting up a trunk? |
02:12.32 | JerJer | uhh - not sure if you can... i haven't noticed a way |
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02:17.33 | fauxalliance | RichardLynch, http://www.voip-info.org/wiki/view/Asterisk+tips+SIP+URI+Dial |
02:20.09 | GTXComm | I am having DTMF issues on IAX trunk. Any suggestions? |
02:20.24 | GTXComm | Works on sip. |
02:21.10 | fauxalliance | GTXComm, dtmfmode=rfc2833 |
02:21.23 | GTXComm | Thanks |
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06:07.43 | p3nguin | fauxalliance: There is no dtmfmode in IAX2. |
06:08.12 | p3nguin | gtxcomm: See above statement to fauxalliance. |
06:09.03 | GTXComm | k |
06:10.35 | GTXComm | Ya, I am still having issues; however, I did get my sip registered, and working the way I know how. It is weird, that it wouldn't take alpha numeric username, for inbound calls. It needed 10 digit phone number for the sip Peer. |
06:11.38 | GTXComm | Therefore, bypassing IAX. Well IAX should work better from what I hear. |
06:12.35 | GTXComm | <---- new here, so I don't really know the rules, if I am chatting too much, please let me know. :) |
06:14.34 | GTXComm | or Free PBX 1.4.X is just not something I am familiar with |
06:16.06 | GTXComm | I am have a free pbx test server that I have hooked up in my co-lo, that I am going to test compatibility with 1.6.x and find out the true issue. |
06:16.37 | GTXComm | for IAX / to IAX |
06:19.38 | GTXComm | Q. Does IAX do (default) rfc2833? I am having a hard time with google, and voip-info on this. |
06:50.31 | Corydon76-home | GTXComm: No, it does not. IAX2 does not do ANY signalling or audio on a port other than 4569 |
06:51.09 | Corydon76-home | GTXComm: this is also why IAX pierces firewalls so effectively |
06:51.55 | Corydon76-home | There's no need for a firewall to understand the protocol in order to route the right ports back; it just works. |
06:52.22 | GTXComm | Thanks |
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07:57.14 | kriegerod | hi all. what can you say about asterisk 1.8 stability? can it be adviced for serious production use? thanks |
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08:39.08 | TehRabbitt | Quick question, what is the difference between extensions.ael and extensions.conf? |
08:39.16 | TehRabbitt | and which one should I be focusing on in 1.8 |
08:39.56 | p3nguin | ael is in, well, ael format... and will get converted to conf internally by asterisk. You should be using conf unless you have a good reason to use something else. |
08:40.10 | TehRabbitt | alrighty thanks :) |
08:43.13 | p3nguin | gtxcomm: There is absolutely no reason whatsoever that you have to use a 10-digit numeric for a device's name. Both SIP and IAX2 configs will allow you to use alphanumeric peer names. Also, if this problem is occurring in FreePBX, you should probably either stop using FreePBX or ask about the problem in the more appropriate FreePBX channel (#freepbx). |
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09:12.07 | TehRabbitt | Hm, trying to get chan-sccp-b installed, but when running "make" on the source, it errors out :-\ any ideas? |
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09:13.16 | fenrus | pastebin the errors |
09:13.25 | fenrus | did you install all dependencies ? |
09:13.39 | TehRabbitt | yes, hold on i'm going to pastebin |
09:14.15 | fenrus | if you are using a packagesysetem you need the -dev packets of the dependecies aswell |
09:14.57 | TehRabbitt | http://pastebin.com/XpvN1jC2 |
09:20.06 | TehRabbitt | any ideas? |
09:22.50 | TehRabbitt | :-\ |
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09:24.42 | TehRabbitt | what causes this btw: |
09:24.43 | TehRabbitt | [Feb 6 04:23:42] WARNING[22666]: db.c:112 dbinit: Unable to open Asterisk database '/var/lib/asterisk/astdb': Permission denied |
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09:28.57 | TehRabbitt | anyone here still :-\ |
09:28.58 | fenrus | TehRabbitt, someone that dont have access to /var/lib/asterisk/astdb is trying to access it. |
09:29.12 | fenrus | "someone" as perhaps the asterisk user or your user ? |
09:29.17 | fenrus | when does this error occur ? |
09:29.25 | TehRabbitt | when i'm logged into the console |
09:29.40 | fenrus | as your user ? |
09:29.43 | TehRabbitt | i do have a phone that is plugged in |
09:29.55 | TehRabbitt | not sure what you mean by as my user, i've just logged into the console as root |
09:30.10 | fenrus | okay |
09:30.24 | fenrus | then it seems that the asteriskuser cant access that file. does it exist? |
09:30.27 | fenrus | are you suing astdb? |
09:30.34 | TehRabbitt | not that i'm aware of |
09:31.07 | TehRabbitt | all I did was download teh 1.8 source, ran make / make install and then went to /etc/init.d/asterisk start... from there i went to the console (asterisk -r) and now i'm getting those messages |
09:31.20 | fenrus | okay |
09:32.20 | fenrus | i would unload the module |
09:32.31 | TehRabbitt | unload what module? i haven't loaded any |
09:32.42 | TehRabbitt | haven't even installed the chan-sccp-b one yet |
09:32.47 | TehRabbitt | it won't compile :-\ |
09:33.00 | fenrus | are you certain that its not built and loaded by default ? |
09:33.19 | fenrus | write module show |
09:33.20 | TehRabbitt | positive, you need to enable it... the .so isn't even listed |
09:33.21 | fenrus | in your console |
09:33.45 | fenrus | func_db.so Database (astdb) related dialplan |
09:34.17 | TehRabbitt | ah you mean that module, perhaps it is enabled by default |
09:34.50 | fenrus | or create the db file and make sure taht asterisk kan read/write to it |
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09:35.45 | TehRabbitt | what is the point of the db file though? |
09:38.41 | TehRabbitt | i made the file and i'm still getting the error :-\ |
09:38.41 | TehRabbitt | [Feb 6 04:37:51] WARNING[22846]: db.c:553 ast_db_gettree: Database unavailable |
09:39.01 | fenrus | theres no database index in the file :D |
09:39.14 | TehRabbitt | how do i make an index 0_o |
09:40.10 | fenrus | i'd try the database -commands in the CLI |
09:40.41 | TehRabbitt | No such command 'database -commands' (type 'core show help database -commands' for other possible commands) |
09:40.57 | fenrus | core show help database |
09:41.43 | TehRabbitt | thre isn't anything there to create an index |
09:43.20 | fenrus | im not even sure that you need an index, but i guess it complains because you gave it a flat file, and it wants some kinda database format |
09:43.32 | fenrus | but if you ask asterisk to create it itself you might get it right |
09:43.41 | fenrus | or disable it if you're not using it. |
09:45.40 | TehRabbitt | disabled it lol |
09:45.55 | TehRabbitt | still can't seem to figure out why i can't complie chan-sccp-b though |
09:47.34 | TehRabbitt | checking if the linker (/usr/bin/ld -m elf_x86_64) is GNU ld... yes checking whether the g++ linker (/usr/bin/ld -m elf_x86_64) supports shared libraries... yes checking for g++ option to produce PIC... -fPIC -DPIC checking if g++ PIC flag -fPIC -DPIC works... yes checking if g++ static flag -static works... yes checking if g++ supports -c -o file.o... yes checking if g++ supports -c -o file.o... (cached) yes checking whe |
09:47.36 | TehRabbitt | whoops |
09:47.42 | TehRabbitt | sorry :-\ |
09:47.48 | TehRabbitt | http://pastebin.com/BHvGpbSR |
09:47.51 | TehRabbitt | meant to send that link |
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10:27.09 | StaRetji | Folks, what could be the problem where I hear signal tone with dahdi, but if I press button on the phone nothing happens. |
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10:27.41 | StaRetji | I can hear key tone, but signal tone do not change, so I can't even call 600 echo even though it was there |
10:28.01 | as001 | hi is it possible to setup moh for radio streaming on station which emits program at 128Kbit/s ? |
10:28.01 | StaRetji | asterisk is set to verbose 15 |
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10:29.43 | as001 | When I set it on 24 kbps station it works great, but when i try on 128 or 64 sound is choppy |
10:29.56 | as001 | I am using 1.6.2.16 |
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10:33.00 | StaRetji | it's almost like dial tones are not recognize, CLI doesn't show anything, as it nothing was pressed. |
10:33.04 | StaRetji | Anyone? :/ |
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11:01.45 | CaseSensitive | does anybody know of a resource with an agi script that can load tagged values in csv format by key ? like (var1=value,var2=value) |
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12:22.02 | verywiseman | how can i hide caller id and my extension number when i make call ? |
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17:05.05 | philippel_mac | ping anyone around who has some experience with the CallCompletionRequest/Cancel app in 1.8 that I could bounce a couple issues off of? |
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20:48.02 | domains25 | hello everyone |
20:52.37 | domains25 | anyone can help with a x100p that does not want to call out to a pstn trunk? |
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21:13.39 | MikeH | can anyone identify what the cause of the following error may be? |
21:13.40 | MikeH | [Feb 6 00:26:50] WARNING[3872]: chan_dahdi.c:2689 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) |
21:13.55 | MikeH | getting terrible echo on connected SIP phones |
21:18.24 | TehRabbitt | when i try to run /etc/init.d/asterisk stop it works, but when I try running start, it doesn't start asterisk, any ideas? |
21:18.49 | b14ck | TehRabbitt: look at /var/log/asterisk/full |
21:18.57 | b14ck | err: /var/log/asterisk/messages sorry |
21:19.01 | TehRabbitt | also for some reason, it keeps trying to run chan_skinny even though I set it for noload => chan_skinny.so |
21:19.31 | TehRabbitt | b14ck: nothing in the messages :-\ |
21:19.36 | b14ck | Then do: |
21:19.47 | b14ck | /etc/init.d/asterisk start; asterisk -r (maybe you'll connect to the term) |
21:19.52 | b14ck | and then see what happens |
21:19.56 | b14ck | sometimes it'lls tartup |
21:19.57 | b14ck | then fail immediately |
21:20.09 | b14ck | but in general, look in all of the /var/log/asterisk/* logs |
21:20.16 | b14ck | it will have errors in there somewhere |
21:20.19 | TehRabbitt | [Feb 6 16:19:33] ERROR[17157]: logger.c:1042 init_logger: Unable to create event log: Permission denied |
21:20.29 | TehRabbitt | i guess that's why |
21:20.33 | b14ck | there you do =p |
21:20.35 | b14ck | *go |
21:20.42 | b14ck | look at /etc/init.d/asterisk |
21:20.46 | b14ck | what USER it is set to run as? |
21:20.57 | TehRabbitt | not sure, how do I set that? |
21:21.06 | b14ck | its a variable defined in /etc/init.d/asterisk |
21:21.09 | TehRabbitt | i'm running the command as root, but i'm not sure if it's actually trying to run as root |
21:21.10 | TehRabbitt | hm |
21:21.16 | b14ck | look at that flie |
21:21.17 | b14ck | *file |
21:21.20 | b14ck | vim /etc/init.d/asterisk |
21:21.37 | TehRabbitt | USER=$NAME GROUP=$USER |
21:21.45 | TehRabbitt | Ah |
21:21.48 | TehRabbitt | NAME= Asterisk |
21:22.43 | TehRabbitt | changed it to root, and now I get this lovely message: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
21:22.57 | b14ck | do: |
21:23.11 | b14ck | /etc/init.d/asterisk stop; /etc/init.d/asterisk start; asterisk -r; |
21:23.29 | TehRabbitt | Grr :-\ looks like the files are all owned by asterisk |
21:23.39 | b14ck | that doesn't matter |
21:23.42 | b14ck | root can read any file |
21:23.45 | TehRabbitt | Starting as root not supported. |
21:23.54 | b14ck | although, if you want to change it, you can do: `chown -R root:root /etc/asterisk` |
21:24.04 | b14ck | How did you install asterisk? And what OS? :o |
21:24.12 | TehRabbitt | Debian, installed 1.6 from source |
21:24.17 | TehRabbitt | logged in as root |
21:24.21 | b14ck | compiled as root? |
21:24.23 | TehRabbitt | yes |
21:24.30 | b14ck | something bad happened in your install somewhere |
21:24.44 | b14ck | can you wipe your box and start fresh? |
21:24.55 | b14ck | I've got some good install instructions if you want them. |
21:24.56 | TehRabbitt | you mean remove asterisk? |
21:25.01 | b14ck | i mean wipe the os |
21:25.05 | b14ck | and start from scratch |
21:25.13 | TehRabbitt | Noooo |
21:25.26 | b14ck | Are you running other services on this box too? :o |
21:25.31 | TehRabbitt | Yes... :-\ |
21:25.35 | b14ck | not good =/ |
21:25.49 | TehRabbitt | well it runs a couple VMs |
21:25.57 | b14ck | well, you can attempt to rm everything |
21:26.11 | TehRabbitt | well thats what i did last time, i had 1.4 installed originally |
21:26.21 | b14ck | rm -rf /etc/asterisk /var/lib/asterisk /etc/init.d/asterisk /etc/init.d/dahdi /var/spool/asterisk |
21:26.36 | TehRabbitt | and then recompile i guess? :-\ |
21:26.44 | b14ck | yah, but follow my instructions |
21:26.50 | b14ck | sec |
21:26.53 | b14ck | http://projectb14ck.org/transparent-telephony-part-2-installing-aster |
21:27.56 | *** join/#asterisk b14ck_ (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
21:28.34 | TehRabbitt | it says for centos |
21:28.46 | TehRabbitt | ah theres an ubuntu one |
21:30.47 | TehRabbitt | hm, would running make menuconfig before running make screw stuff up? |
21:32.26 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
21:34.08 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
21:37.44 | *** join/#asterisk superbeef (~Lane.Jenn@74-120-209-234-rev.redanvil.net) |
21:38.20 | TehRabbitt | ok, i've installed asterisk using source, ran the make install, and now when I try to do /etc/init.d/asterisk start it says it doesn't exist :-\ |
21:38.45 | superbeef | you'll have to make your own init.d script |
21:38.51 | superbeef | it uses a script called safe_asterisk |
21:38.55 | superbeef | which is basically a loop |
21:38.58 | TehRabbitt | how would I do that? |
21:39.06 | superbeef | what os |
21:39.10 | TehRabbitt | debian |
21:39.13 | TehRabbitt | squzeeze |
21:39.19 | TehRabbitt | squeeze* |
21:40.49 | superbeef | here's one i use on ubuntu |
21:40.51 | superbeef | shoudl be close enogh |
21:40.52 | superbeef | http://pastebin.com/cSrJ035v |
21:41.03 | superbeef | modify it as needed and add it with rc-update or whatever it is that debian uses |
21:41.54 | TehRabbitt | -bash: /etc/init.d/asterisk: Permission denied |
21:42.08 | superbeef | ewps |
21:42.13 | superbeef | that script isn't so good anyway because that's for freepbx |
21:42.20 | TehRabbitt | lol |
21:42.46 | superbeef | most people |
21:42.49 | superbeef | jsut run safe_asterisk |
21:43.08 | TehRabbitt | and how do i do that thoguh |
21:43.19 | superbeef | and add /usr/sbin/safe_asterisk into /etc/rc.local |
21:43.38 | superbeef | just run it from the shell |
21:43.42 | superbeef | should be in your path |
21:43.45 | superbeef | "safe_asterisk" |
21:43.55 | TehRabbitt | ah it works |
21:48.52 | TehRabbitt | so what is safe_asterisk in comparison to asterisk? |
21:49.59 | drmessano | It's a script that runs asterisk |
21:50.28 | drmessano | http://linux.die.net/man/8/safe_asterisk |
21:50.33 | TehRabbitt | drmessano it seems like there is nothign iniside /etc/asterisk anymore :-\ |
21:51.24 | TehRabbitt | where would the .conf files be? |
21:51.57 | TehRabbitt | hm nvm it's there, weird |
21:52.06 | drmessano | O.o |
21:52.28 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.200.16.55) |
21:52.30 | DelphiWorld | hi guys |
21:52.31 | TehRabbitt | i did a cd /etc/asterisk and it said it couldn't be found... i did just cd /etc then cd asterisk and it worked 0_o |
21:52.33 | DelphiWorld | here we go aguin. |
21:52.41 | DelphiWorld | module iax2 is unable to be loaded for me |
21:52.44 | DelphiWorld | always same issue |
21:52.52 | DelphiWorld | i'm sure i'm missing something else. any help? |
21:53.41 | superbeef | techrabbit.. did you do make samples |
21:53.51 | superbeef | do install example config scripts |
21:55.04 | TehRabbitt | yea they're there now |
21:55.27 | TehRabbitt | for some reason the files were there, maybe i mispelt it or something 0_o |
21:55.35 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
21:55.46 | TehRabbitt | now i just have to find out why it's rejecting my phone :-\ |
21:56.05 | DelphiWorld | superbeef: my issue is that sip and iax channel isn't loaded :( |
21:56.22 | superbeef | not loaded or not configured? |
21:56.39 | DelphiWorld | superbeef: you you rock rock |
21:56.43 | DelphiWorld | superbeef: make samples fixed it |
21:56.58 | superbeef | word |
21:58.33 | TehRabbitt | how can I confirm the module is installed? |
21:58.40 | TehRabbitt | for chan_sccp-b |
21:59.05 | DelphiWorld | TehRabbitt: module load chan_you.so :-P |
21:59.26 | TehRabbitt | yea just figured that out 0_o lol and yet another error... i swear it can never just *work* lol |
21:59.27 | TehRabbitt | [Feb 6 16:58:57] WARNING[1610]: loader.c:428 load_dynamic_module: Error loading module 'chan_sccp': /usr/lib/asterisk/modules/chan_sccp.so: undefined symbol: ast_rtp_change_source |
21:59.45 | Kobaz | do you need sccp? |
21:59.52 | TehRabbitt | [Feb 6 16:58:57] WARNING[1610]: loader.c:781 load_resource: Module 'chan_sccp' could not be loaded. |
21:59.56 | TehRabbitt | Kobaz: yes I do |
22:00.10 | Kobaz | okay so, the easy way isn't possible (just not loading it) |
22:00.14 | TehRabbitt | lol |
22:00.38 | Kobaz | sccp probably has a dependency on another module |
22:00.52 | Kobaz | either that, or it's broken in your asterisk version |
22:01.11 | TehRabbitt | http://chan-sccp-b.sourceforge.net/ |
22:01.14 | TehRabbitt | thats what i'm trying to use |
22:01.22 | TehRabbitt | i'm using the latest trunk via SVN |
22:02.34 | Kobaz | no guarantees that trunk will work at all |
22:02.47 | TehRabbitt | Ah |
22:02.52 | TehRabbitt | so go for one of the stable ones i guess? |
22:03.01 | Kobaz | yes, most definitly |
22:03.28 | TehRabbitt | aight, so how should I go about removing the old module :-\ |
22:03.39 | TehRabbitt | or would running make clean do it? |
22:03.42 | Kobaz | wipe out /var/lib/asterisk/modules |
22:04.00 | TehRabbitt | and reinstall asterisk again? |
22:04.20 | Kobaz | a stable asterisk version |
22:04.41 | TehRabbitt | well it's just the chan-sccp-b one that i got from svn |
22:04.47 | TehRabbitt | asterisk is 1.6 via the site |
22:04.54 | DelphiWorld | use 1.8 |
22:05.11 | TehRabbitt | can't use 1.8 with chan_sccp-b :-\ it just doesn't compile the module |
22:06.14 | DelphiWorld | TehRabbitt: contact cisco :-D |
22:06.37 | TehRabbitt | DelphiWorld: Funny... lol |
22:06.57 | DelphiWorld | :) |
22:07.52 | *** join/#asterisk Dr-Linux (~Dr-Linux@182.177.131.100) |
22:08.15 | Dr-Linux | why should i go to 1.6.2 and not for 1.8.x? |
22:08.29 | Dr-Linux | should go for* |
22:28.00 | *** join/#asterisk Faithful (~Faithful@carame.lnk.telstra.net) |
22:40.05 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
22:50.48 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
23:03.59 | DelphiWorld | if i see a peer unmonitored |
23:04.02 | DelphiWorld | how do i make it monitored? |
23:04.07 | DelphiWorld | puzzled: :) |
23:06.47 | DelphiWorld | changed from peer to friend... |
23:09.15 | Tozz_ | add qualify=yes to the peer |
23:09.17 | Tozz_ | in sip.conf |
23:10.51 | DelphiWorld | Tozz_: is iax... but i allready got it |
23:10.54 | DelphiWorld | thank you Tozz_ |
23:10.56 | Tozz_ | ah ok |
23:10.57 | Tozz_ | np |
23:11.43 | *** join/#asterisk jql (~jql@12.9a.344a.static.theplanet.com) |
23:12.42 | DelphiWorld | Tozz_: now i'm stuck in iax registration, is not registering itself to the provider |
23:13.37 | Tozz_ | u hgave a register => line? |
23:14.12 | DelphiWorld | Tozz_: yeah and unable to receyv call :( |
23:15.29 | IsUp | 'iax2 show registry' on your CLI |
23:18.48 | DelphiWorld | IsUp: iax2 show registry isn't this show iax2 client registered to my asterisk? |
23:19.16 | DelphiWorld | IsUp: zero iax2 reg... |
23:21.52 | *** join/#asterisk infobot (ibot@rikers.org) |
23:21.52 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.2.3 (2010/01/26), 1.6.2.16.1 (2010/01/18), 1.4.39.1 (2010/01/18), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
23:22.28 | DelphiWorld | strange. |
23:22.43 | IsUp | DelphiWorld: you have register => in your iax conf, right? |
23:22.54 | DelphiWorld | yes, IsUp |
23:23.06 | IsUp | so you are trying to register to your provider, |
23:23.30 | IsUp | you need to use 'iax2 show registry' |
23:23.36 | IsUp | try 'iax2 reload' first |
23:24.16 | DelphiWorld | IsUp: yeah, registering to my providers |
23:24.30 | DelphiWorld | IsUp: still zero iax2 reg |
23:25.05 | IsUp | can you post your iax.conf to pastebin please? |
23:25.15 | IsUp | mask any passwords before posting |
23:26.41 | shortcircuit | Scenario: I have VoicePulse connecting PSTN to my Asterisk server with a static IPv4 address. I have reliable 6to4 on the same server, and reliable 6to4 at home. Can Asterisk/SIP work directly, IPv6 to IPv6? It saves me setting up and managing a proxy SIP gateway on my router. |
23:27.04 | DelphiWorld | IsUp: iax.conf or only the peer? |
23:27.43 | IsUp | DelphiWorld: whole iax.conf |
23:29.13 | DelphiWorld | IsUp: here is my peer http://dpaste.de/UyaG/ |
23:29.25 | DelphiWorld | IsUp: while give iax2.conf... iax3 ;-) |
23:30.33 | IsUp | DelphiWorld: make sure your register => is under [general] context, not a peer |
23:30.45 | IsUp | also, do 'core set verbose 10' |
23:30.50 | IsUp | and 'iax2 reload' |
23:31.42 | IsUp | if you cant see any on 'iax2 show registry', do 'iax2 set debug on' and then 'iax2 reload' |
23:31.56 | DelphiWorld | IsUp: http://dpaste.de/UUqL/ |
23:32.00 | DelphiWorld | IsUp: is my iax.conf |
23:32.58 | IsUp | go to line 212, put your 'register => user:pass@london.voip.ms' there |
23:33.54 | DelphiWorld | IsUp: i did... let see |
23:34.20 | DelphiWorld | wow, IsUp |
23:34.30 | MikeH | hrm |
23:34.34 | DelphiWorld | IsUp: why isn't registering me using my peer definition and is working in line 212? |
23:34.58 | IsUp | register => definition must be under [general] context |
23:35.10 | MikeH | is it considered secure to have a SIP port open to the internet? |
23:35.24 | DelphiWorld | MikeH: if no, avoid using VoIp. |
23:35.26 | IsUp | you can do #include "iax_registrations.conf" on your line 212 |
23:35.40 | MikeH | DelphiWorld? |
23:35.58 | DelphiWorld | MikeH: is sure, but you have to take care about your PBX security |
23:36.13 | MikeH | DelphiWorld, This is my concern |
23:36.29 | MikeH | I have no issues with opening it up - all accounts have passwords |
23:36.31 | DelphiWorld | IsUp: :) |
23:36.31 | MikeH | and are fairly secure |
23:36.33 | DelphiWorld | IsUp: good catch |
23:36.44 | MikeH | but the concern is still there |
23:37.04 | MikeH | perhaps I'll use SSH tunelling |
23:37.23 | IsUp | DelphiWorld: ;) |
23:38.04 | IsUp | MikeH: sip scanning is going wild, they can hammer your server. i prefer close 5060 to strangers. |
23:38.13 | IsUp | or use fail2ban if your 5060 open |
23:39.02 | MikeH | I'll use ssh tunnelling I think :) |
23:40.28 | IsUp | i dont know how it works but lemme say again, always keep your 5060 closed! |
23:40.51 | DelphiWorld | MikeH: ssl/tls |
23:42.05 | MikeH | also |
23:42.12 | MikeH | can anyone recommend a windows based SIP client? |
23:42.47 | DelphiWorld | MikeH: phoner |
23:43.11 | IsUp | MikeH: xlite/eyebeam |
23:44.04 | DelphiWorld | IsUp: exten => 883510009900637,1,Echo() |
23:44.06 | DelphiWorld | IsUp: is this correct? |
23:44.23 | IsUp | DelphiWorld: yes, but you dont need () |
23:46.19 | DelphiWorld | IsUp: hanging up directly |
23:46.27 | DelphiWorld | exten => 883510009900637,1,Answer() |
23:46.27 | DelphiWorld | exten => 883510009900637,1,Echo |
23:47.39 | IsUp | DelphiWorld: can you see any output on CLI when call hits your PBX? |
23:47.50 | DelphiWorld | yeah, IsUp... i see :-) |
23:49.11 | DelphiWorld | -- Call accepted by 78.129.153.20 (format ulaw) |
23:49.11 | DelphiWorld | -- Format for call is (ulaw) |
23:49.11 | DelphiWorld | -- Hungup 'IAX2/voipms-18460' |
23:49.17 | DelphiWorld | IsUp: this is the log |
23:49.21 | DelphiWorld | leifmadsen: don't ban me :-) |
23:50.08 | DelphiWorld | IsUp: ok, IsUp! when thay will be down? :-) |
23:50.22 | IsDown-no-IsUp | IsDown-no-IsUp: :P |
23:50.27 | IsDown-no-IsUp | IsUp: :) |
23:51.12 | DelphiWorld | i think while give up on it now and go to sleep, is 1am |
23:51.46 | IsUp | yeap i am back |
23:51.53 | IsUp | working or not? |
23:53.30 | DelphiWorld | IsUp: no, not working :-) |
23:53.49 | MikeH | IsUp, What is the "Domain" field on x-lite? |
23:54.18 | IsUp | IP of PBX |
23:54.49 | IsUp | DelphiWorld: paste your iax.conf to pastebin again, please |
23:55.04 | DelphiWorld | spsure |
23:55.06 | DelphiWorld | IsUp: sure |
23:56.01 | DelphiWorld | IsUp: http://dpaste.de/166A/ |
23:56.17 | IsUp | DelphiWorld: and to 'core set verbose 10' and place a call to your DID and then paste output to pastebin |
23:56.45 | IsUp | DelphiWorld: i cant see voip.ms in your iax.conf |
23:57.01 | DelphiWorld | IsUp: because i do include :-) |
23:58.13 | IsUp | paste your provider context too |
23:58.52 | DelphiWorld | IsUp: i did put it in default |
23:59.06 | DelphiWorld | IsUp: log: http://dpaste.de/aRJP/ |