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| 08:02.29 | tiny | Hi. How would I send local echo to my headphones? I'm talking to a SIP client which is only speaker without microfone. I would like to hear what I'm saying in my headphones. |
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| 08:34.49 | marksm2 | Hi, do you know if Asterisk 1.4.42-BRIstuffed-0.4.0-RC16 has some vulnerabilities. If I open the SIP 5060 port to the public internet, I have people successfully using my BRI ISDN (PSTN) using SIP without having having a valid login on the server (I use long password). Could it be a vulnerability or a dial plan misconfiguration? Any docs where I can check that my dialplan is secure against intruders? thanks |
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| 08:47.03 | mzdravkov | Can I get events from ARI without using a Stasis application |
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| 08:58.50 | mzdravkov | What I want is to be able to connect to ARI and get events whenever a dial is made, without changing my dialplan and calling some stasis application everywhere in it. |
| 09:00.08 | defsdoor | from ARI ? |
| 09:00.59 | mzdravkov | yes |
| 09:01.38 | mzdravkov | I'm not sure whether I understand this correctly, I'm completely new to asterisk |
| 09:02.02 | defsdoor | Isn't using AMI easier ? |
| 09:02.28 | defsdoor | in this sense anyway as you are just watching |
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| 09:04.15 | Jesterboxboy | i have a problem with asterisk recognizing dtmfnumbers double, the dtmf is sent using rfc2833, but i have the impression that somehowe asterisk also detects inband signal? is there a way to turn inband tmf detection off? |
| 09:04.15 | mzdravkov | defsdoor: oh, maybe you are right - I'll read about AMI, but on prima vista it looks like it's more appropriate |
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| 09:19.28 | Jesterboxboy | if i enter dtmfmode= in sip.conf, does that mean, we will send dtmf to that sip peer in that mode? or do we expect dtmf from that peer in that mode? |
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| 10:32.14 | marlinc | I'm getting pjsip_evsub_register_pkg: Assertion `mod_evsub.mod.id != -1' failed. |
| 10:32.25 | marlinc | While running Asterisk, any have idea what could cause this? |
| 10:35.06 | file | it can happen if all the PJSIP modules aren't built against a shared library version of PJSIP, or if the res_pjsip_pubsub module isn't loaded but other modules which use it are |
| 10:37.38 | marlinc | Disabling the res_pjsip_refer.so module did solve the problem but now I'm getting ../src/pj/os_core_unix.c:1232: pj_mutex_lock: Assertion `mutex' failed. |
| 10:37.39 | marlinc | mm |
| 10:37.47 | marlinc | 13:34:47.111 sip_endpoint.c !Processing incoming message: Request msg REGISTER/cseq=33534 (rdata0x909c5c4) |
| 10:37.52 | marlinc | Was the message before that |
| 10:38.02 | marlinc | The strange thing is that this install worked fine on another host |
| 10:38.11 | marlinc | We've moved this OpenVZ container from one host to another |
| 10:38.36 | file | pastebin the output of /usr/lib/asterisk/modules/res_pjsip.so |
| 10:38.38 | file | er |
| 10:38.43 | file | ldd /usr/lib/asterisk/modules/res_pjsip.so |
| 10:38.48 | file | is still half asleep |
| 10:39.16 | marlinc | file, https://gist.github.com/Marlinc/404037e472b4e66b978f |
| 10:42.02 | file | the PJSIP modules were built against a static version of PJSIP |
| 10:42.14 | file | as a result each module has a self contained copy of it, which will not work at all |
| 10:43.42 | marlinc | Ouch, the be honest I've got no idea what to do about that. Could you possibly guide me in the right direction? |
| 10:43.42 | file | PJSIP will need to be rebuilt, as will Asterisk |
| 10:44.03 | marlinc | Well I have done that about 5 minutes ago. I guess I have to change some flags? |
| 10:44.06 | file | instructions are on the wiki at https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject |
| 10:44.44 | marlinc | It probably has to do with --enable-shared? |
| 10:44.50 | file | yes |
| 10:45.07 | marlinc | I just did ./configure |
| 10:45.32 | file | that would be why, --enable-shared is not a default |
| 10:46.19 | marlinc | I did the same on Asterisk itself, do I have to add any flags to Asterisks configure? |
| 10:46.43 | file | no |
| 10:46.59 | file | you will likely need to do a "make distclean" on both and start fresh |
| 10:47.11 | file | and if you add a --prefix to pjsip that does not match where it was installed originally |
| 10:47.15 | file | you will end up with two PJSIP installs on your system |
| 10:47.25 | file | so either match it, or leave the --prefix out |
| 10:47.59 | marlinc | What's the difference between make clean and make distclean? |
| 10:48.27 | file | distclean wipes configure stuff as well, it is as if you downloaded a fresh copy |
| 10:48.34 | file | clean only removes the built stuff |
| 10:49.07 | marlinc | Ah okay, the actual module configuration will stay? By that I mean the modules that I have selected using menuselect |
| 10:49.47 | file | that will likely disappear as well - you can just do a make clean and see if it works |
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| 10:52.47 | marlinc | The next reinstall of this PBX will use the Ubuntu packages. It does however mean we'll have to use a non LTS |
| 10:56.03 | marlinc | I'm still getting the same error file. I have used make distclean |
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| 10:58.52 | file | did you re-run configure in Asterisk? |
| 10:59.24 | marlinc | Yes I did |
| 11:00.12 | marlinc | I have now ran make clean and make uninstall and removed all libraries (related to PJ) I could find while running "ldconfig -p | grep pj" |
| 11:00.26 | file | updatedb and locate libpj |
| 11:00.29 | file | will find everything |
| 11:01.47 | marlinc | It appears I got some libraries in /usr/lib/lib |
| 11:01.49 | marlinc | As well |
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| 11:04.23 | marlinc | There we go, everything has been removed |
| 11:04.40 | marlinc | The line shown in the guide should be fine file? So ./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr CFLAGS='-O2 -DNDEBUG' |
| 11:04.42 | marlinc | Including the prefix? |
| 11:04.51 | file | yes |
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| 11:04.54 | marlinc | Good |
| 11:04.56 | mjamilfarooq | anybody to help mme out |
| 11:04.59 | mjamilfarooq | ? |
| 11:06.28 | file | mjamilfarooq, if you ask a question someone might answer |
| 11:06.58 | mjamilfarooq | i have video support enable in sip.conf and also has allow=ulaw allow=h263p I want receiver to know somehow if caller is calling audio or audio/video both |
| 11:08.21 | mjamilfarooq | basically i can't differentiate at receiver end if call is audio call or audio/video call |
| 11:10.00 | mjamilfarooq | both audio and audio/video calls have audio/video status true. I can also see this in SIP/SDP |
| 11:11.34 | mjamilfarooq | even though at caller channel in extensions.conf I can read audionativeformat/videonativeformat to determine codec of caller channel |
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| 11:13.44 | mjamilfarooq | anybody? |
| 11:21.36 | mjamilfarooq | if somebody doesn't understand what i am trying to say I can elaborate further |
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| 11:26.17 | marlinc | file, do you know what could cause 'Unable to create request with auth.No auth credentials for any realms in challenge.'? We haven't had that before |
| 11:26.37 | marlinc | Registration works fine. It happens trying to make a call |
| 11:27.10 | marlinc | Ah, I think I know what the problem is |
| 11:27.26 | marlinc | It has to do with our VoIP provider |
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| 11:30.50 | mjamilfarooq | expert opinion required |
| 11:37.27 | marlinc | Its working, thanks a lot file |
| 11:39.30 | file | marlinc, nifty |
| 11:44.17 | mjamilfarooq | file, has i asked something really stupid not to get a reply |
| 11:44.40 | file | people are not obligated to provide a response, they may not know |
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| 14:35.11 | glNito | suddenly I can' |
| 14:35.23 | glNito | I cant run service asterisk start |
| 14:36.13 | [TK]D-Fender | Verify that it isn't already running. Then start it manually and see what happens |
| 14:37.10 | glNito | asterisk is an unrecognized service |
| 14:37.23 | glNito | Strange. |
| 14:37.27 | [TK]D-Fender | Sounds like your init script isn't there... |
| 14:38.16 | glNito | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 14:38.17 | glNito | [root@sip1 /]# Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 14:38.23 | glNito | oops, didn't mean for that |
| 14:46.14 | igcewieling1 | glNito: "make config" in the Asterisk source directly. |
| 14:47.22 | igcewieling1 | glNito: "asterisk -cvvv" should give you more information on why Asterisk isn't starting. |
| 14:48.31 | [TK]D-Fender | Actually... make sur to add "g" and specify the user&group, etc |
| 14:48.48 | [TK]D-Fender | Otherwise you'll end up missing things like permissions issues, etc |
| 14:48.53 | glNito | I ran asterisk -vvvvvvc and it came up just fine |
| 14:49.00 | igcewieling1 | [TK]D-Fender: good point |
| 14:49.15 | [TK]D-Fender | glNito: Now do it better. |
| 14:49.34 | igcewieling1 | /etc/asterisk/asterisk.conf should also contain the username to run as |
| 14:52.53 | [TK]D-Fender | Or CLI parameters |
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| 15:07.37 | mrmccrac | I have X-Lite registering okay with SIP, in the packet it includes the port in the From: portion of the packet. in asterisk my register => line also include the port, but when viewing the packet in wireshark, the From: is not including the :port |
| 15:08.00 | mrmccrac | as a result the remote SIP server is sending me a 404 not found |
| 15:08.58 | mrmccrac | anyone know how to tell asterisk to include that? |
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| 15:46.02 | mrmccrac | think i finally figured it out |
| 15:47.35 | Phil-Work | can I dial from the Asterisk CLI on Asterisk 11? |
| 15:47.42 | Phil-Work | found some stuff on Google but it's all invalid commands |
| 15:47.48 | Phil-Work | "originate", "dial" |
| 15:47.58 | igcewieling | Phil-Work: "core show applications" |
| 15:49.02 | Phil-Work | how do I trigger the Dial application from the CLI? |
| 15:49.43 | igcewieling | Phil-Work: you do not. |
| 15:49.59 | igcewieling | I believe you would use originate |
| 15:54.07 | Phil-Work | ah, got it |
| 15:54.09 | Phil-Work | channel originate |
| 15:54.10 | Phil-Work | thanks |
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| 16:14.10 | glNito | of course everything goes wrong on Friday |
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| 16:24.52 | drab | hi there, looks like I'm confused about permit/deny in sip.conf. I have a very basic asterisk 11 configured and all phones get ips from dhcp in a specific ip range. I wanted to restrict asterisk to those ips only and so in sip.conf I added permit=192.168.6.0/255.255.255.0 deny=0.0.0.0/0.0.0.0 |
| 16:25.14 | drab | but then I can register and call from a softphone on 192.168.10.0/24 |
| 16:25.40 | [TK]D-Fender | PB it |
| 16:26.32 | [TK]D-Fender | (your config that is) |
| 16:26.35 | [TK]D-Fender | ~pb |
| 16:26.36 | 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. |
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| 16:27.07 | ODUQUE | Hola! alguien habla español? |
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| 16:28.28 | ODUQUE | hola alguien habla español por favor? |
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| 16:30.46 | drab | http://pastebin.com/6KC2dZJC |
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| 16:31.20 | drab | so I'd expect 998 to not be able to register anymore after adding those permit/deny and not being able to call out to 999 (or 999 being able to call 998 either) |
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| 16:37.36 | ghost75 | somebody knows how i can see dnsmgr output on console? |
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| 16:39.37 | ghost75 | its enabled but is not showing what its doing |
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| 16:40.46 | ghost75 | and is dnsmgr issuing sip reregister on ip change? |
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| 16:50.51 | mub | So my boss gave me some prototype PCIe timer card because he doesn't like our USB ones. There's nothing written on the card or packaging... lspci doesn't seem to have it listed. |
| 16:51.08 | mub | I assume there's some driver to compile, but I have no idea how I'm going to find it |
| 16:52.06 | mub | Should I just use that kernel module that makes it so we don't require a hardware timer and just pretend that I'm using this thing? |
| 16:52.27 | mub | That's what I'm going to do, thanks guys :D |
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| 18:30.15 | Blackthorn | I'm trying to install freepbx on my xen server, and upon installing the o.s. it keeps stopping on package 517 of 625. it's called Freepbox-12.0.45-1.shmz I thought perhaps i didn't provide enough disk space but up up to 40megs now. I'm using the template for Centos 7 Thoughts |
| 18:30.21 | Blackthorn | ? |
| 18:37.49 | mub | Blackthorn: How much diskspace? My newer freepbx system (~30 users, couple voicemails) is using 7.9G |
| 18:38.19 | mub | Also, if it was running out, it would probably alert you before attempting to install any packages |
| 18:38.39 | Blackthorn | i've got it set at 40gig, however, the package finally went through adn it's continuing with the install.. just wierd it took so long on that package |
| 18:41.48 | Blackthorn | You think 20gig should be plenty for a asteirsknow install? |
| 18:43.58 | [TK]D-Fender | yes, though I'd have gone with the official FreePBX disto vs that |
| 18:44.06 | [TK]D-Fender | or DIY |
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| 18:52.53 | Blackthorn | I'm using an older asterisk system and I do understant the sip and dialplan config files. Whats going to be the easiest for me to transition into? (i'm not an experianced amin of *nix but my old system does run on centos and i've recnetly build some centos dns servers. I want to stick with centos as well |
| 18:55.37 | [TK]D-Fender | Depends what you want |
| 18:55.43 | [TK]D-Fender | if You're going to DIY.. then DIY |
| 18:55.54 | [TK]D-Fender | if you're going to stick with a GUI then go for the FreePBX official distro. |
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| 19:10.59 | mub | I really want to build an asterisk system and learn how to configure it completely without a WUI/GUI. I would learn so much. |
| 19:11.33 | mub | I would be a dialplan master |
| 19:15.58 | robmal | Is that a synonym for NotGettingLaid black belt? |
| 19:22.09 | mub | robmal: I'll have you know that I *could* get sex if I did some chores around the house |
| 19:22.17 | mub | I have to earn it, but I don't wanna! |
| 19:22.39 | mub | I'd rather build asterisk :D |
| 19:24.23 | robmal | Whatever floats your boat. Install debian, download ast13, compile it, run it, make a softphone connect to it, get a sip trunk, connect to it, make a call. |
| 19:24.27 | robmal | Or do heroin. |
| 19:24.33 | robmal | ~Trainspotting. |
| 19:28.04 | mub | heroin would be nice if it wasn't fucking $20 for .1 |
| 19:28.26 | robmal | So asterisk it is. |
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| 19:31.35 | [TK]D-Fender | is going to catch that dragon any day now.... |
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| 19:55.33 | ghost75 | why everybody keeps saying sip trunks? |
| 19:56.07 | robmal | Because [TK]D-Fender gets mad. He has elephant issues. |
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| 20:25.27 | learath | Anyone gotten conferencing working on Cisco phones? |
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| 20:34.53 | MaliutaLap | you wouldn't like [TK]D-Fender when he gets mad - because he backs up his arguments with well researched reasoned facts. |
| 20:35.57 | robmal | But i have proof of trunks in the internet. |
| 20:36.01 | robmal | https://www.google.com/trends/explore#q=sip%20trunk%2C%20sip%20connection |
| 20:36.03 | robmal | Observe. |
| 20:36.34 | robmal | There are about 8-9 times more trunks than connections. |
| 20:36.37 | MaliutaLap | robmal: I would, but it hurts when IP |
| 20:36.46 | robmal | ;-) |
| 20:37.50 | robmal | And now i found a proof the trunks are strong. |
| 20:37.52 | robmal | https://www.google.com/trends/explore#q=sip%20trunk%2C%20sip%20connection%2C%20sip%20peer%2C%20voip%20provider&cmpt=q&tz=Etc%2FGMT-2 |
| 20:38.03 | robmal | There are more trunks than voip providers. |
| 20:38.55 | robmal | Well, at least there're more people looking for them so it looks like in 2010 every one found their voip provider and now there are only trunks left. |
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| 20:50.16 | lanning | ??? The difference between a SIP connection and a SIP trunk is ????? zipo, ziltch, nada |
| 20:55.12 | lanning | The definition of "trunk" has been "a collection of channels" like in a T1 with 24 channels would be a trunk line. |
| 20:55.45 | lanning | SIP is a registration with no concept of a channel. |
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| 21:00.42 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:00.54 | [TK]D-Fender | lanning, If even... |
| 21:02.18 | lanning | True. The registration is not a requirement either. (only for dynamic endpoints) |
| 21:02.48 | lanning | It is just a negotiation protocol to start data streams |
| 21:03.32 | lanning | Of course, it does kind of say that in the name... Session Initiation Protocol |
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| 21:32.08 | mrmccrac | anyone know how to configure asterisk to use rport/rinstance when doing invites? |
| 21:33.08 | learath | Does anyone have a working example of waiting for an extension with Background/WaitExten? |
| 21:34.35 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 21:39.23 | [TK]D-Fender | call background... call waitexten. |
| 21:39.38 | [TK]D-Fender | Have extens in there for them to dial |
| 21:39.40 | [TK]D-Fender | The end. |
| 21:39.50 | learath | [TK]D-Fender: I'm doing that. it takes the first digit, then freaks out because the first digit does not match anything |
| 21:40.11 | [TK]D-Fender | then you should look at what you thought you made |
| 21:40.29 | learath | uhm. I'm clearly missing something. |
| 21:40.39 | learath | How do I tell it to actually wait for multiple digits? |
| 21:40.58 | [TK]D-Fender | By having a pattern in there that actually even needs one |
| 21:41.19 | [TK]D-Fender | because if it freaks out on the first.. you didn't do that much correctly yet |
| 21:41.45 | learath | I think I do? ( exten => _1NXXXXXXXXX,n,Dial(SIP/stuff) |
| 21:41.59 | [TK]D-Fender | as written? no |
| 21:42.09 | [TK]D-Fender | PASTEBIN is your friend |
| 21:42.12 | [TK]D-Fender | ~pb |
| 21:42.13 | 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. |
| 21:42.15 | [TK]D-Fender | ^^^ |
| 21:42.33 | learath | hmm. |
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