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| 01:51.02 | UncleKiwi | Hi, I have a client getting echo on calls when using an ATA caller can hear them selves |
| 01:51.22 | UncleKiwi | i know normally there is some echo cancelation in the cisco devices |
| 01:51.38 | UncleKiwi | but this is a grandstream 24 port channel bank |
| 01:51.40 | UncleKiwi | thingy |
| 01:52.31 | UncleKiwi | can asterisk deal with echo cancellation, or is it best to find a way to sort it at the ATA |
| 01:52.39 | UncleKiwi | it does not happen with a hardphone |
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| 02:44.10 | ChannelZ | If it's an ATA->SIP then no asterisk doesn't deal with echo cancellation. It really should happen on the ATA's side |
| 02:46.41 | UncleKiwi | yeah its seems this device has a bit of echo |
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| 03:50.02 | UncleKiwi | I seem to get the echo and im talking to the person using the phone attached to the ATA attached to the PBX |
| 03:50.22 | UncleKiwi | and im on my own PBX with a proper hardphone |
| 03:50.37 | UncleKiwi | but i seem to be able to hear myself echo |
| 03:50.41 | UncleKiwi | as I talk |
| 04:02.05 | UncleKiwi | i think i fixed it it was transcoding |
| 04:06.46 | UncleKiwi | i had the calls comming in on g722 over sip |
| 04:07.03 | UncleKiwi | and the channel bank was talking ulaw |
| 04:07.18 | UncleKiwi | this seemed to cause echo |
| 04:07.52 | [TK]D-Fender | Transcoding has nothing to do with echo. |
| 04:08.04 | [TK]D-Fender | Echo is at the point of TDM conversion |
| 04:09.44 | UncleKiwi | can you explain why its stopped echoing ? |
| 04:09.52 | UncleKiwi | after i made them all talk ulaw |
| 04:11.54 | [TK]D-Fender | Shouldn't be relavent to any change |
| 04:23.03 | UncleKiwi | seems odd because the echo is gone |
| 04:41.17 | UncleKiwi | the person attached to the ATA was not complaining about the echo - its the people they called that coul hear echo |
| 04:41.29 | UncleKiwi | and im using a raspberrypi b+ |
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| 08:08.58 | sudharshaw | Hi I have an analog card in my asterisks box which has been running with out a problem for more than 2 years, I am using asterisk 1.4 recently when I take a call the call doesnot go through |
| 08:09.30 | sudharshaw | in the log I see app_dial.c: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented) |
| 08:10.07 | sudharshaw | I researched the net and found that this could be due to the zap module not being load in to asterisks |
| 08:10.23 | sudharshaw | none of the zap commands are available in the asterisk CLI |
| 08:10.53 | sudharshaw | so I tried restarting the zaptel service and found out that there was an issue when ztcfg is running |
| 08:10.59 | sudharshaw | on one of the channels |
| 08:11.21 | sudharshaw | so I #commented that line and restarted the service and it started fine |
| 08:11.34 | sudharshaw | but I still don't see the chan_zap loaded in asterisks |
| 08:11.51 | sudharshaw | when I do a lsmod I can see that zaptel module is loaded |
| 08:12.01 | sudharshaw | does any one has a clue? |
| 08:12.32 | sudharshaw | I also can't find chan_zap.so in the modules directory |
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| 08:48.16 | gustopn | hi folks |
| 08:49.16 | gustopn | does anyone have an idea if asterisk is able to change codecs while on line with someone? |
| 08:50.17 | gustopn | i have the impression that I locked out myself by using directrtpsetup=yes |
| 08:50.57 | gustopn | I did read something about this being the problem of the setting that asterisk is then not able to adjust the SDPs any more. because it does not forward it? |
| 08:52.29 | gustopn | I have a VoIP provider who supports G.722 and it works quite well, as long as someone calls me. when I want to call someone, it goes G.722 out from my side, but he is sending me first the info about how much the call is and bullshit like this, and this is always G.711 |
| 08:54.09 | gustopn | So I have then a connection that is G.722 up and G.711 down, maybe it changes, however, it gets lost and I worked out the theory that it might be the case thats because the SDP is not comming through, however, to investigate further, I intend to collect some logs on this over the weekend |
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| 11:20.48 | stefan27 | in dialplan is $[ "YES" = "yes" ] true? |
| 11:21.50 | WIMPy | Why should it? |
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| 11:25.06 | stefan27 | it shouldn't but it could |
| 11:29.10 | gustopn | hi WIMPy |
| 11:29.16 | gustopn | WIMPy: how are you? |
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| 13:42.59 | *** join/#asterisk hayonj (4077d488@gateway/web/freenode/ip.64.119.212.136) |
| 13:44.39 | hayonj | hi can anyone help me choose the proper module to install for a basic installation of Asterisk ? I'm following this tutorial http://wiki.freepbx.org/display/HTGS/Installing+FreePBX+12+on+CentOS+6.5#InstallingFreePBX12onCentOS6.5-Installiksemel and I'm at this step Compile and install Asterisk > make menuselect |
| 13:46.41 | [TK]D-Fender | ALL of them |
| 13:47.41 | [TK]D-Fender | Take everything you can get. |
| 13:48.46 | hayonj | [TK]D-Fender: Thanks again mate ! You're a champion for always being here and advising people ! |
| 13:48.54 | [TK]D-Fender | You're welcome |
| 13:49.39 | hayonj | [TK]D-Fender: Yet with all the hacks, going around and the CVE's out there I wanted to limit the scope of what the system can do. |
| 13:51.09 | [TK]D-Fender | hayonj: The only things that can be hacked are outside interfaces (channel drivers, AMI, etc). |
| 13:51.19 | [TK]D-Fender | Those you can selectively DISABLE in modules.conf |
| 13:51.25 | [TK]D-Fender | if you really feel you have to. |
| 13:51.46 | [TK]D-Fender | But if you end up needing them you can tehn REENABLE them without having to go through the mess of trying to compile what you're missing after the fact |
| 13:51.59 | hayonj | [TK]D-Fender: What about deprecated modules ? like for example ... cdr_mysql ? Wasn't it that one I was using to check my Call Records Database while using the Elastix version ? |
| 13:52.01 | [TK]D-Fender | SAve yourself the trouble by doing it immediately and then fine tune what gets used |
| 13:52.26 | [TK]D-Fender | If it's there... then your system supports it. Use what you want from the rest. |
| 13:53.55 | newtonr | hayonj, if it helps, there is a fairly minimal modules.conf set that comes with the source now - configs/basic-pbx/modules.conf - it is intended for use with the Super Awesome Company basic pbx configs. It serves as a good example of a minimal explicitly loaded module set. |
| 13:55.11 | newtonr | That is of course running Asterisk without FreePBX. I'm sure you'll probably need a lot more for that. |
| 13:55.18 | hayonj | I didn't even know about this file modules.conf ... installing from source and communicating here really gives me a better idea on how to tackle an install for optimal security ! |
| 13:56.33 | [TK]D-Fender | hayonj: Basically the only ones you might want to get rid of are channel drivers you aren't using. |
| 13:56.48 | newtonr | hayonj, we haz documentation https://wiki.asterisk.org/wiki/display/AST/Configuring+the+Asterisk+Module+Loader |
| 13:58.06 | hayonj | [TK]D-Fender: can you give me the whole list of external interfaces and how to secure the system properly ? (If this all works out properly I'll write a guide with my steps) |
| 13:58.50 | [TK]D-Fender | hayonj: basically if you don't need IAX2, H323, etc, then NOLOAD those modules out. |
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| 13:59.04 | [TK]D-Fender | hayonj: lok at "chan_XXX.so" in your modules folder |
| 14:01.17 | hayonj | ok, i'll install everything as per your suggestion [TK]D-Fender ... including deprecated modules |
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| 14:03.20 | hayonj | newtonr: Thanks for the link for the modules page |
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| 14:04.05 | GlenK | Does fxotune work in the normal way when the PBX is using a gateway device? |
| 14:04.16 | newtonr | hayonj, np |
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| 14:05.26 | hayonj | Can I go without DAHDI ? |
| 14:07.23 | hayonj | No one uses traditional analog phones ... |
| 14:07.46 | hayonj | I'll just install everything and disable the module |
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| 14:10.12 | hayonj | [TK]D-Fender: Utilities > Extended > astman is disabled ... isn't that a module I need ? I've seen that in my previous installs. |
| 14:11.26 | [TK]D-Fender | hayonj: No point to go without it. Some things still depend on it. DO IT |
| 14:11.35 | [TK]D-Fender | (DAHDI that is) |
| 14:11.43 | hayonj | [TK]D-Fender: say, I compile eveything by default, and just click next ... if I ever need something I might have forgotten, how do I recompile |
| 14:11.58 | [TK]D-Fender | Same way you're compiling NOW |
| 14:13.10 | hayonj | so /usr/src/asterisk* make menuselect ? and go from that step ... |
| 14:16.41 | hayonj | [TK]D-Fender: back to hacking throught external interfaces, I'm looking at this picture https://wiki.asterisk.org/wiki/display/AST/Asterisk+Architecture%2C+The+Big+Picture so what you were talking about is the first 2 layers, network and hardware ? other than that they can't get through ? is that right ? |
| 14:17.30 | [TK]D-Fender | hayonj: Only thing they can hack is something listening on a network interface really... |
| 14:18.12 | [TK]D-Fender | hayonj: Which pretty much means AMI, ARI(?), and Channel drivers. |
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| 14:27.22 | hayonj | [TK]D-Fender: the only way I've been succesful recently is change the default ports, for example SSH instead of 22 it would be 12323 mysql 3306 would be 12324 and the httpd deamon would be off if not needed, say I don't need to add an extension or a follow-me .... I would simply do service httpd stop, the attacks stopped since then, but really I think the httpd stop really helped |
| 14:33.12 | hayonj | [TK]D-Fender: I added make samples and make progdocs but I get this error [root@ASTERISK asterisk-13.3.2]# make progdocs # Note, Makefile conditionals must not be tabbed out. Wasted hours with that. Doxygen is not installed. Please install and re-run the configuration script. |
| 14:33.15 | [TK]D-Fender | hayonj: typicaly MySQL should only be accessed directly by your server itself, never an outside client. At that point you'd FIREWALL it so packets can't even arrive |
| 14:33.35 | [TK]D-Fender | hayonj: Also changing that one specifically can make other things that assume the default freak out. |
| 14:34.03 | hayonj | [TK]D-Fender: I installed i Doxygen ... using yum install still the same error ... |
| 14:34.42 | hayonj | [TK]D-Fender: they say re-run the config file ... which is it ? the ./configure or the make menuselect ? |
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| 14:38.08 | [TK]D-Fender | not sure what error you're referring to... |
| 14:38.11 | hayonj | tight now, I have it as is on a deployed installation [root@ASTSCI2 ~]# netstat -tulpn | grep mysql tcp 0 0 0.0.0.0:3306 0.0.0.0:* LISTEN 2394/mysqld |
| 14:39.07 | hayonj | [TK]D-Fender: was reffering to that : +---- Asterisk Installation Complete -------+ + + + YOU MUST READ THE SECURITY DOCUMENT + + + + Asterisk has successfully been installed. + + If you would like to install the sample + + configuration files (overwriting any + + existing config files), run: + + |
| 14:39.31 | [TK]D-Fender | ~pb |
| 14:39.35 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 14:39.35 | hayonj | <PROTECTED> |
| 14:39.38 | [TK]D-Fender | ^^^ |
| 14:39.47 | [TK]D-Fender | I don't see the actual error. |
| 14:39.51 | [TK]D-Fender | PASTEBIn this stuff.... |
| 14:39.56 | hayonj | ok |
| 14:41.23 | hayonj | [TK]D-Fender: http://pastebin.com/841xqGJB |
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| 14:50.52 | marceloamorim | hello guys, anyone knows some book about nat + voip that covers everything? |
| 14:51.25 | GlenK | nat? as in having a router or whatever do some forwarding? |
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| 14:56.30 | hayonj | [TK]D-Fender: sorry that was a noob question, I just re-ran the ,/configure make menuselect make make install and then make progdocs .... it worked ... |
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| 15:48.47 | dillbilly | i i thought i had resolved an issue i had yesterday where an incoming call on my pri would get answered and is supposed to get re-directed back out on the pri, but i'm getting an all circuits busy message again. our provider is using an ascending order on the pri and directed us to configure asterisk to use descending. it looks like sometimes it's still trying to call back out on pri channel 0 for some reason http://pastebin.com/1aZdVAm2 http://pas |
| 15:48.48 | dillbilly | tebin.com/DxLR2fBA |
| 15:50.32 | dillbilly | my pri card supports 24 channels, and we have 10 on our circuit, so i broke out the channels into g0 with 10 channels and g1 with 13, and only use g0 in an outbound route, which is set to 10 channels max. |
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| 15:51.44 | boch | hello |
| 15:52.19 | dillbilly | not g0 & 1, group 0 and 1 |
| 15:53.09 | dillbilly | dahdi g0 is set to use pri group 0 descending |
| 15:53.23 | boch | do you know if is it possible to cancel a transfer if called extension does not answers? |
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| 16:06.26 | hayonj | [TK]D-Fender: I just finished installing from sources. I can't get to the administration GUI. tcp 0 0 :::80 :::* LISTEN 32525/httpd should it not listen to 443 by default ? |
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| 17:45.32 | marceloamorim | Guys, I'm getting this warning, but I don't have any phone registered on this asterisk yet. =:::>>>> WARNING[684]: chan_sip.c:4110 retrans_pkt: Timeout on 0ff65f0761980d27bd9def448e926e99 on non-critical invite transaction. |
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| 17:55.19 | [TK]D-Fender | hayonj: No, the default configuration tends to listen on HTTP, not HTTPS |
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| 18:40.20 | WIMPy | dillbilly: Your log sasys you tried to call an invalid number. |
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| 18:41.01 | yang | tzafrir: hi ! |
| 18:47.40 | tzafrir | yang, hi |
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| 18:58.37 | michaely | I am working on adding support for Asterisk 13 and PJSIP to our AMI based application. One requirement that we have is to allow auto answer on SIP devices for the initial call back during execution of an OriginationAction, via the AMI. When using chan_sip we accomplish this by adding SIPADDHEADERX variables to the Origination action, but this does not seem to work with PJSIP. I have also tried PJSIPADDHEADERX variables with no luck. Does anyone know how |
| 18:58.37 | michaely | <PROTECTED> |
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| 19:00.02 | dillbilly | WIMPy, where do you see that? i see several errors at pastebin lines 44-49, 137-138, and then the final set when it hangs up at 231-253, but i'm not understanding quite what the issue is. i'm using pbx in a flash and have it set to call a misc destination if the user is unavailable, and i get the circuits error every time if i route through an extension, but for a while if i set the DID destination and the misc destination the call would go through. |
| 19:00.58 | WIMPy | I see hangupcause 1. |
| 19:01.46 | WIMPy | And if you want to know exactely what's going on, enable pri debug. |
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| 19:14.12 | cmendes0101 | Hey, Anyone try playing a file with unicode in the filename? Was going to try but thought I'd ask |
| 19:23.57 | [TK]D-Fender | dillbilly: I'm betting that is counting as LONG DISTANCE and you have to dial the 1 in front |
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| 19:46.17 | dillbilly | curious. i'll look into that. |
| 19:54.11 | dillbilly | nope |
| 19:59.23 | marceloamorim | could you point a direction for me to fix this warning about chan_sip.c timeout on non-critical invite transaction? |
| 20:05.51 | *** join/#asterisk iamnotarobot (~iamnotaro@gateway/vpn/privateinternetaccess/iamnotarobot) |
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| 20:20.19 | [TK]D-Fender | marceloamorim: fix your networking /NAT config |
| 20:24.07 | marceloamorim | but there is none sip peers |
| 20:24.24 | [TK]D-Fender | huh? |
| 20:24.35 | WIMPy | Ah |
| 20:24.52 | WIMPy | It's normal when someone tries to hack you. |
| 20:25.05 | [TK]D-Fender | Nothing is normal yet |
| 20:25.19 | [TK]D-Fender | we don't see WHO or WHAT that is related to at all |
| 20:25.23 | *** join/#asterisk iamnotarobot (~iamnotaro@gateway/vpn/privateinternetaccess/iamnotarobot) |
| 20:25.27 | [TK]D-Fender | Because he isn't actually SHOWING us the call |
| 20:25.34 | [TK]D-Fender | Or describing anything with any detail |
| 20:25.50 | marceloamorim | there is no call, its just the asterisk |
| 20:26.03 | [TK]D-Fender | that is cleary for a call |
| 20:26.21 | [TK]D-Fender | Asterisk is not throwing warnings out of thin air |
| 20:26.26 | [TK]D-Fender | it is trying to talk to something |
| 20:26.29 | [TK]D-Fender | and you aren't LOOKING |
| 20:26.33 | marceloamorim | like WIMPy said, maybe someone trying to rack, because there is none sip and call running |
| 20:26.41 | *** join/#asterisk iamnotarobot (~iamnotaro@gateway/vpn/privateinternetaccess/iamnotarobot) |
| 20:26.50 | [TK]D-Fender | THERE IS A SIP CALL |
| 20:26.56 | [TK]D-Fender | lOOK AT THE DEBUG |
| 20:27.20 | marceloamorim | so its a ghost call |
| 20:27.23 | marceloamorim | nice |
| 20:27.28 | WIMPy | Yes. There is (or was) a call. |
| 20:27.28 | [TK]D-Fender | no |
| 20:27.41 | [TK]D-Fender | That is an ACTUAL communication and you aren't looking at the debug |
| 20:27.50 | [TK]D-Fender | We do not know who or what it is just yet |
| 20:27.53 | [TK]D-Fender | because you are not LOOKING |
| 20:27.55 | [TK]D-Fender | GO LOOK |
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| 20:29.19 | [TK]D-Fender | packs up to head home |
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| 20:30.16 | Sudravirodhin | Curious if anyone else has run into this: Asterisk 13, attended transfers do not honor attempts to change caller ID via dialplan. Blind transfers obey caller ID changes, though. |
| 20:30.49 | Sudravirodhin | I used DumpChan() to confirm CallerIDNum was not changed. |
| 20:31.30 | marceloamorim | now I put debug on my cli and I'm getting worker thread idle timeout reached. dying |
| 20:31.42 | WIMPy | Before or after transfer? |
| 20:32.54 | Sudravirodhin | @WIMPy before and after show wrong Caller ID in DumpChan() as if Set() had no effect. |
| 20:33.35 | WIMPy | i'm not sure what you're up to. Up until the transfer is completed, it's just a normal call. |
| 20:33.52 | WIMPy | And once it is completed, there's no dialplan involved. |
| 20:35.09 | Sudravirodhin | My goal is to show Caller A's (A -> B, B -> C) number to Caller C. Once I start the feature ("transfer" is played) it ignores the statements that set its Caller ID as it Dial()s. |
| 20:35.42 | WIMPy | That will happen automatically. |
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| 20:36.04 | Sudravirodhin | Via CLID, yes, but I'm looking to have that done prior to transfer completion. |
| 20:36.50 | WIMPy | Oh, you're using feature transfers? |
| 20:37.00 | Sudravirodhin | Yes, is there an atxfer function? |
| 20:37.26 | WIMPy | On your phone, hopefully. |
| 20:38.26 | Sudravirodhin | Afraid not. I guess it is as grim as I thought. |
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| 20:41.31 | Sudravirodhin | Thanks for the reply, WIMPy. Appreciate it. |
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| 20:45.12 | Sudravirodhin | Vicidial handles attended transfers itself, right? |
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| 20:46.38 | svm_invictvs | Is it possible to program asterisk to send an outging call, once it connects, have it call another person? |
| 20:47.26 | WIMPy | Yes, Originate: Via *CLI, call files or AMI. |
| 20:47.27 | robmal | Yes. |
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| 20:57.06 | [TK]D-Fender | marceloamorim |
| 20:57.06 | [TK]D-Fender | now I put debug on my cli and I'm getting worker thread idle timeout reached. dying <- you are still clearly not looking at SIP DEBUG |
| 20:57.18 | [TK]D-Fender | marceloamorim, Those warnings are not "debug" |
| 20:57.31 | [TK]D-Fender | marceloamorim, "sip set debug on" <------- |
| 20:58.42 | marceloamorim | yeah, I found the son of !@#*& was trying to register on my brand new asterisk |
| 20:59.23 | marceloamorim | thx to you guys |
| 21:00.19 | marceloamorim | sometimes the asterisk don't show the complete information I used to |
| 21:01.21 | marceloamorim | because when I try to connect and I don't have permission or I'm missing the password, the asterisk show me that way, this guys was doing all this without I notice |
| 21:02.59 | marceloamorim | this is the first time I try to debug this way to find some IP |
| 21:03.02 | marceloamorim | Via: SIP/2.0/UDP 195.154.154.110:5070;branch=z9hG4bK-8ad65ad367c550cd597fc141681033c9;received=195.154.154.110;rport=5070 |
| 21:03.11 | marceloamorim | ty all |
| 21:05.03 | marceloamorim | I'll take my break |
| 21:05.57 | [TK]D-Fender | hops in the shower |
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| 22:04.29 | svm_invictvs | hrm |
| 22:04.35 | svm_invictvs | Okay, so let's say I wanted to do this. |
| 22:05.08 | svm_invictvs | I dial in to an extension, I dial a number hit pound, then dial another number hit pound... |
| 22:05.24 | svm_invictvs | The outgoing Caller ID is set to the first number, and dials the second number. |
| 22:05.36 | svm_invictvs | So I can dial out of my office # without revealing my cell phone # |
| 22:05.39 | svm_invictvs | Is that possible? |
| 22:07.11 | Sudravirodhin | Should be. Using Read() twice, then Set(CallerID(num)=${FIRST}) then Dial() ${SECOND}, etc. |
| 22:07.16 | Sudravirodhin | My idea, anyway. |
| 22:08.21 | svm_invictvs | yeah |
| 22:14.04 | svm_invictvs | http://mysticpaste.com/view/uZuOTZlFD7;jsessionid=18zo2nnqsa6fl5ixpg1o881tl?2 |
| 22:14.06 | svm_invictvs | Something like that? |
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| 22:22.34 | Sudravirodhin | Basically. |
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| 22:23.35 | Sudravirodhin | Haven't quite used Read myself yet but I'm working on a context for use with GoSub and features' applicationmap. |
| 22:23.44 | Sudravirodhin | Hope it works for you. |
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| 22:43.30 | svm_invictvs | So Set(CALLERID(all)="Foo Caller <5555551212>"); doesn't seem tow do anything. |
| 22:43.48 | svm_invictvs | The receipient of the call jsut gets the caller ID |
| 22:43.53 | svm_invictvs | I wonder if my provider is blocking that. |
| 22:43.57 | svm_invictvs | They didn't used to. |
| 22:45.11 | Sudravirodhin | I avoid using (all) because I have mixed results. |
| 22:45.17 | svm_invictvs | hrm |
| 22:45.20 | svm_invictvs | Just num? |
| 22:45.23 | Sudravirodhin | (num) and (name) separately work for me and my provider. |
| 22:45.32 | Sudravirodhin | But (num) should work solo. |
| 22:45.44 | svm_invictvs | WHo's your provider, if I may ask? |
| 22:45.48 | svm_invictvs | I'm using Teliax |
| 22:46.05 | svm_invictvs | I knew they were pretty unscrupulous |
| 22:48.22 | Sudravirodhin | Telnyx |
| 22:48.39 | Sudravirodhin | You should contact your provider to see if they can help you. |
| 22:48.49 | Sudravirodhin | I find phone calls (ironically) help considerably. |
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