| 00:00.06 | [TK]D-Fender | [19:58]Razvawell the idea is that after Dial it should be forwarded to...the SIP company, who should call the phone... <- what "SIP company"? |
| 00:00.40 | Razva | which I've configured in the iax.conf, I suppose |
| 00:00.52 | [TK]D-Fender | Razva: And what have you fogotton to do? |
| 00:01.18 | Razva | tooo link the dial to the iax? |
| 00:01.25 | [TK]D-Fender | CORRECT! |
| 00:01.51 | [TK]D-Fender | And if they are IAX ... they are not SIP. |
| 00:01.57 | [TK]D-Fender | Which is another inversion... |
| 00:02.15 | [TK]D-Fender | (IAX2 technically) |
| 00:02.56 | raspberrypifan | any idea why i get this |
| 00:02.56 | raspberrypifan | root@raspbx:~# raspi-config |
| 00:02.56 | raspberrypifan | Segmentation fault |
| 00:03.37 | Razva | aaaaaaaaa, I should dial IAX not SIP? |
| 00:03.57 | [TK]D-Fender | Razva: what is the protocol you configured for it? |
| 00:04.16 | johndropper | I try to dial into a meetme conf for example and this is what I get [2014-05-13 20:03:29] WARNING[1153][C-00000003]: chan_sip.c:10543 process_sdp: We are requesting SRTP for audio, but they responded without it! |
| 00:04.36 | Razva | iax but the provider sent me sip (into the activation mail) |
| 00:04.47 | johndropper | this is extention 121 trying to dial 500 |
| 00:04.49 | Razva | so I should somehow add the provider into the sip.conf also? |
| 00:05.01 | [TK]D-Fender | Then that's what you should configure if they have not said either will work with their info |
| 00:06.09 | johndropper | If anyone can help that would be awesome |
| 00:06.12 | Razva | got it |
| 00:06.16 | johndropper | this one has me stumped |
| 00:06.21 | Razva | so I've moved the credentials from IAX to SIP |
| 00:06.31 | Razva | but now I should tell same => n,Dial(SIP/${EXTEN}) to dial that provider |
| 00:06.46 | johndropper | Maybe [TK]D-Fender can help |
| 00:06.47 | Razva | which I really have no idea of how to do :| |
| 00:07.04 | johndropper | he seems super smart |
| 00:07.14 | [TK]D-Fender | [20:06]Razvabut now I should tell same => n,Dial(SIP/${EXTEN}) to dial that provider <--- how does that dial KNOW what provider to use? |
| 00:07.32 | [TK]D-Fender | Razva: You could ahve configure a THOUSAND in sip.conf |
| 00:08.05 | [TK]D-Fender | waits for the light |
| 00:08.09 | Razva | wait wait, is EXTEN? |
| 00:08.23 | Razva | ot |
| 00:08.25 | [TK]D-Fender | that holds the number you ar currently processing in the dialplan. |
| 00:08.31 | [TK]D-Fender | that is the NUMBER... not where it GOES |
| 00:08.40 | [TK]D-Fender | almost there..... |
| 00:09.22 | Razva | dial -> protocol SIP -> number |
| 00:09.38 | Razva | ooooh 0 intuition! :)) |
| 00:09.54 | Razva | I need to tell him to dial the second provider in the sip |
| 00:10.17 | [TK]D-Fender | Dial(Tech/peer/number) |
| 00:10.33 | Razva | ahaaaaaaaa |
| 00:10.37 | [TK]D-Fender | ther is no "second provider". There is only the one you explicity type right into that command |
| 00:10.54 | [TK]D-Fender | you tell it where to go in the dial directly. No guess, not automatic |
| 00:11.15 | Razva | SIP/MyProvider/${EXTEM} ? |
| 00:11.24 | [TK]D-Fender | for sip, sure |
| 00:11.30 | [TK]D-Fender | And fixing the typo, etc |
| 00:12.00 | viasanctus | can the cisco spa303 use a centralized address book with asterisk? |
| 00:12.08 | Razva | [Netmaster] < the "title" of the entry in the sip.conf |
| 00:12.11 | Razva | so it should be like |
| 00:12.23 | Razva | n,Dial(SIP/Netmaster/${EXTEN}) ? |
| 00:13.39 | [TK]D-Fender | You seem to be getting it now... |
| 00:13.41 | Razva | YEEEEEEEEEEEEEES |
| 00:13.42 | Razva | YEEEEEEEEES |
| 00:13.45 | Razva | it's ringing |
| 00:13.47 | [TK]D-Fender | very good |
| 00:13.47 | Razva | woooooooooooohooooooooooo |
| 00:13.52 | Razva | PARTYYYYYYYYYY |
| 00:14.14 | Razva | boom chicka boom :D |
| 00:14.46 | Razva | well...after all...it wasn't...so complicated |
| 00:14.50 | Razva | :D |
| 00:15.01 | Razva | now I finally understand...doooh... |
| 00:16.00 | [TK]D-Fender | A worthwhile process. the flow concepts are very important. the dialplan is 90%+ of Asterisk |
| 00:16.18 | Razva | question: can it show a name, not a number? |
| 00:16.37 | [TK]D-Fender | A few SIP peers entered into a config = no big deal. actually processing calls and doing specific stuf.. that's the real work |
| 00:16.52 | [TK]D-Fender | There is a CALLERID name .... AND a number |
| 00:17.06 | [TK]D-Fender | Now which if either you are able to actually set... depends on your provider |
| 00:17.35 | Razva | weeeeeeeell...now it shows 100 |
| 00:17.42 | Razva | 100 which would be the username |
| 00:21.30 | viasanctus | dialing # shows the shared address book right? |
| 00:21.35 | viasanctus | on all sip phones? |
| 00:21.52 | viasanctus | is that an option to be supported by the telephone? |
| 00:22.22 | WIMPy | no, no, possibly |
| 00:22.25 | viasanctus | And I can create multiple address books in freepbx, how does the phone know in which address to look or which caller name to display? |
| 00:22.26 | [TK]D-Fender | There is no general concept of a phonebook like that |
| 00:22.32 | viasanctus | ow |
| 00:22.37 | viasanctus | i've read it in a blog |
| 00:22.39 | Razva | [TK]D-Fender interesting thing. if I rename 100 with a name (ex. Razva) it'll give "wrong password" |
| 00:22.45 | [TK]D-Fender | And this isn't the place for FreePBX config advice..... |
| 00:22.53 | viasanctus | guessing it's asterisk.. |
| 00:22.58 | [TK]D-Fender | it isn't |
| 00:23.08 | Razva | so auth should be done only based on a number? |
| 00:23.19 | [TK]D-Fender | did you change your softphone as well? |
| 00:23.23 | Razva | yup |
| 00:23.32 | viasanctus | freepbx isn't an asterisk management application? |
| 00:23.37 | [TK]D-Fender | perhaps didn't apply all your changes properly |
| 00:23.57 | viasanctus | http://en.wikipedia.org/wiki/FreePBX FreePBX is an open source GUI (graphical user interface) that controls and manages Asterisk (PBX) |
| 00:25.20 | WIMPy | ... and is not part of Asterisk. |
| 00:25.29 | WIMPy | ~freepbx |
| 00:25.29 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 00:25.56 | *** join/#asterisk bmurt (~brendan@64-121-18-195.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 00:27.11 | viasanctus | it doesn't really matter if i need to access the asteirsk config files directly or go through freepbx |
| 00:27.17 | Razva | solved it, I suppose it was a typo. :| |
| 00:27.28 | Razva | ok, so if I do callerid=Webhipo, than the caller on the phone is "Unknown" |
| 00:27.44 | Razva | if I do callerid="Webhipo" <123456> the caller is 123456 |
| 00:27.55 | viasanctus | and to be honest, It's nothing compared to enterprise softwae development we are involved (core of our business) |
| 00:27.57 | [TK]D-Fender | viasanctus: Asterisk has no phonebook concept. |
| 00:28.02 | viasanctus | ok thank you :) |
| 00:28.04 | Razva | my desire is to have the caller "Webhipo" |
| 00:28.18 | [TK]D-Fender | viasanctus: Nor do any common VoIP protocols, etc |
| 00:28.21 | viasanctus | i find asterisk one of the most simplest things around tbh |
| 00:28.32 | viasanctus | just no experrience |
| 00:28.53 | viasanctus | thank you Fender! |
| 00:29.19 | [TK]D-Fender | viasanctus: If you want to know how to do it for a phone... read the phone's manual to see if there is an option at all. |
| 00:29.31 | viasanctus | a personal address book is described |
| 00:29.32 | [TK]D-Fender | viasanctus: But it'll be specific to that model |
| 00:30.01 | viasanctus | I do read "Your personal address book is a place to store contact information about people you frequently call and people who are not included in your corporate directory, such as clients, suppliers, friends, and family members. " |
| 00:30.03 | viasanctus | hence my q |
| 00:30.31 | [TK]D-Fender | And what else does it say as to where a corporate directory might be held? |
| 00:30.41 | Razva | [TK]D-Fender can you please give me a hint of how to accomplish that also? and than I promice I'll not bother you in the next 24 hours! :D |
| 00:30.43 | viasanctus | need to figure out how to attach a corporate directory :) but I get it now, asterisk is about the voice management, no side services related with user management |
| 00:30.51 | viasanctus | fender, nothing |
| 00:31.07 | viasanctus | have been googling my butts of for the last 1,5 hour |
| 00:31.14 | johndropper | anyone? |
| 00:31.34 | [TK]D-Fender | http://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/spa303-3-line-ip-phone/qa_c67-613606.html |
| 00:31.51 | [TK]D-Fender | viasanctus: 2 second search on my part seemed to turn up instant solid answers... |
| 00:32.02 | [TK]D-Fender | https://www.google.ca/#q=cisco+SPA+303+corporate+directory |
| 00:32.05 | [TK]D-Fender | FIRST result |
| 00:32.13 | viasanctus | fender, sorry, you didn't get my q |
| 00:32.26 | viasanctus | I know it can handle LDAP |
| 00:32.39 | johndropper | any help with my srtp error? |
| 00:32.48 | viasanctus | but I was wondering if there's something with asterisk (built in) I know now asterisk doesn't ,so perhaps freepbx |
| 00:33.04 | viasanctus | well |
| 00:33.05 | viasanctus | lol |
| 00:33.05 | johndropper | WARNING[1153][C-00000007]: chan_sip.c:10543 process_sdp: We are requesting SRTP for audio, but they responded without it! |
| 00:33.10 | viasanctus | I do now freepbx does |
| 00:33.22 | viasanctus | i was looking faultive since I didn't know where to draw the line |
| 00:34.09 | *** join/#asterisk zerohalo (~zerohalo@2601:6:f80:224:9999:c971:6a89:f097) |
| 00:36.36 | johndropper | Honestly I would exspect anyone to know |
| 00:36.58 | johndropper | asterisk meesed up big time on this. Its a 11.6 to 11.9 difference |
| 00:37.10 | johndropper | it has to do with rtp encrption’ |
| 00:37.20 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 00:37.31 | [TK]D-Fender | gah |
| 00:37.42 | [TK]D-Fender | missed everything since my last line... |
| 00:37.47 | *** join/#asterisk jhester (~jhester@64.134.221.244) |
| 00:37.47 | johndropper | my phones are set as RTP ENCRYTION ON. When I disable RTP encrytion on the phones every works |
| 00:38.12 | johndropper | my prob is I dont want to go around to 65 plus phones and disable rtp encryp |
| 00:38.36 | johndropper | so I am wondering how I can get asterisk to just work with my phones rtp encryption on |
| 00:39.40 | [TK]D-Fender | Set up the * to match |
| 00:40.17 | johndropper | TK what? |
| 00:40.48 | [TK]D-Fender | If your phones are using SRTP then Asterisk has to be configured to MATCH them |
| 00:41.03 | johndropper | nope worked in 1.6 all day long |
| 00:41.19 | johndropper | oops meant 11.6 |
| 00:41.23 | johndropper | 11.9 killed me |
| 00:41.28 | [TK]D-Fender | Something clearly is not right with the config. |
| 00:41.38 | johndropper | the config of what? |
| 00:41.40 | Razva | [TK]D-Fender ok, so if I do callerid=Webhipo, than the caller on the phone is "Unknown". if I do callerid="Webhipo" <123456> the caller is 123456. my desire is to have the caller "Webhipo". can you please give me a hint of how to accomplish that also? and than I promice I'll not bother you in the next 24 hours! :D |
| 00:41.44 | [TK]D-Fender | either |
| 00:42.03 | johndropper | either |
| 00:42.07 | [TK]D-Fender | if I do callerid=Webhipo, <- where? |
| 00:42.20 | Razva | sip.conf |
| 00:42.28 | johndropper | op |
| 00:42.41 | [TK]D-Fender | Razva: callerid = "name" <number> |
| 00:43.14 | Razva | got it, but is there any way to "force" it to use the name? |
| 00:43.17 | Razva | not the number |
| 00:44.10 | [TK]D-Fender | "it"? |
| 00:44.23 | [TK]D-Fender | You almost never get to set a NAME when calling the PSTN |
| 00:44.39 | [TK]D-Fender | So if you'r thinking of pushing that out VoicePulse.... don't hold your breath |
| 00:44.53 | [TK]D-Fender | For one of your SIP pones to another... sure |
| 00:45.32 | johndropper | anyone smart enough? |
| 00:45.34 | Razva | I'm using some local provider, Netmaster |
| 00:45.38 | *** join/#asterisk jhe5ter (~jhester@64.134.221.244) |
| 00:45.56 | Razva | so this is a limitation from the provider? |
| 00:45.58 | [TK]D-Fender | johndropper: Asterisk and your phone do not agree |
| 00:46.04 | [TK]D-Fender | johndropper: Fix them so they do |
| 00:46.10 | johndropper | lol |
| 00:46.15 | [TK]D-Fender | Razva: this is a limitation of almost all |
| 00:46.16 | johndropper | you dont even know |
| 00:46.21 | johndropper | great answer |
| 00:46.25 | johndropper | lol |
| 00:46.32 | [TK]D-Fender | johndropper: You can change either side so that it agrees with the other |
| 00:46.38 | [TK]D-Fender | Which one YOU want to do is up to you |
| 00:46.44 | johndropper | ok how do I change the asterisk side |
| 00:46.51 | [TK]D-Fender | vi sip.conf |
| 00:46.52 | johndropper | maybe disable srtp all together |
| 00:46.58 | johndropper | Im there |
| 00:47.00 | johndropper | been there |
| 00:47.10 | [TK]D-Fender | If your phones ask for and asterisk can't deliver your calls will fail. |
| 00:47.20 | [TK]D-Fender | You seem to be missing the part about making them "agree" |
| 00:47.32 | johndropper | no it worked great in 11.6 |
| 00:47.36 | johndropper | 11.9 killed them |
| 00:47.47 | [TK]D-Fender | So either set up Asterisk properly for SRTP, or disable it in your phones. |
| 00:48.00 | johndropper | i cant disable in in the phones. |
| 00:48.03 | johndropper | not an option |
| 00:48.09 | johndropper | So how in asterisk |
| 00:48.12 | [TK]D-Fender | Apparently you can... and as you said.,... you just don't want to |
| 00:48.15 | johndropper | maybe disable srtp |
| 00:48.24 | johndropper | I did on 1 phone to test the theory |
| 00:48.34 | [TK]D-Fender | And it worked as you said |
| 00:48.36 | [TK]D-Fender | so you can |
| 00:48.40 | johndropper | Im not going to drive to every location and change |
| 00:48.53 | [TK]D-Fender | Well don't waste time lying to us at least |
| 00:48.55 | johndropper | this is a hosted asterisk server |
| 00:49.09 | [TK]D-Fender | If you want to look at a phone... then look at a phone. |
| 00:49.11 | johndropper | each phone connected is 100 miles away |
| 00:49.13 | Razva | [TK]D-Fender any idea of a provide who allows this...? I basically want to call clients from time to time, only when they *really* need my call, but without showing my number. If I call from a restricted (unknown) number most of them will not answer, so I was thinking to call them with the company name so they will answer... :| |
| 00:49.40 | johndropper | so should I drive 100 miles to each phone and change |
| 00:49.42 | johndropper | no |
| 00:49.43 | johndropper | so |
| 00:49.48 | johndropper | Now what sir? |
| 00:49.52 | [TK]D-Fender | Razva: you might be able to have VoicePulse SET the name for your calls. |
| 00:50.07 | Razva | sounds interesting. |
| 00:50.47 | Razva | ok than, thanks for your time! |
| 00:51.36 | [TK]D-Fender | You're welcome |
| 00:51.50 | Razva | sent you a PM. :) |
| 00:53.00 | jameswf | If im going to be 100 miles from a phone I may invest in one with a web interface |
| 00:53.21 | johndropper | yeah james great ideaswf |
| 00:53.53 | johndropper | so let me remote into 65 phones and spend 1 month doing that |
| 00:54.51 | [TK]D-Fender | could issue config changes and force all to reboot remotely right from his server.... if he ever needed to |
| 00:55.00 | jameswf | If I have 65 phones to manage I may invest in an endpoint manager |
| 00:55.24 | [TK]D-Fender | jameswf: Min'es called "vi". It's like magic and stuff! |
| 00:55.49 | johndropper | I am in VI |
| 00:55.52 | johndropper | now what? |
| 00:55.58 | johndropper | again I say |
| 00:56.14 | johndropper | you must not know. I am vi sip.conf tell me what to do smarty |
| 00:56.27 | [TK]D-Fender | I never said that would help with the way you set yours up. |
| 00:56.34 | [TK]D-Fender | You are making assumptions |
| 00:56.41 | [TK]D-Fender | And being snarky about it |
| 00:56.42 | johndropper | 11.6 works great 11.9 NO |
| 00:56.47 | jameswf | [TK]D-Fender vi is the devil I dont care what you say lalalalalala |
| 00:56.55 | johndropper | lol |
| 00:56.57 | [TK]D-Fender | yeah, and that version magically breaks everything |
| 00:57.02 | johndropper | it did |
| 00:57.24 | johndropper | I went back to 11.6 and what do you know. it works again .hhhhhmmmmm |
| 00:57.52 | jameswf | Stick to 11.6 is there something pressing you need in 11.9 |
| 00:57.54 | johndropper | so obviously 11.9 has munked something up with rtp encryption |
| 00:58.00 | johndropper | patches |
| 00:58.18 | [TK]D-Fender | Or maybe there is something about things like the USERAGENT that changed with the upgrade so that previous comms don't match' |
| 00:58.20 | johndropper | I like to stay fresh consdering this server is sitting on the public interent |
| 00:58.28 | [TK]D-Fender | Then again you have not shown anything useful in debugging this |
| 00:58.42 | johndropper | I showed error |
| 00:58.58 | [TK]D-Fender | You think that is enough to see where the difference is? |
| 00:59.01 | [TK]D-Fender | 1 little line |
| 00:59.11 | johndropper | WARNING[1152][C-00000002]: chan_sip.c:10543 process_sdp: We are requesting SRTP for audio, but they responded without it! |
| 00:59.13 | [TK]D-Fender | Is that you idea of "debugging" and "complete information"? |
| 00:59.24 | johndropper | lol |
| 00:59.33 | johndropper | give advice. Im listening |
| 00:59.39 | johndropper | ill try it |
| 01:00.24 | [TK]D-Fender | We'll start the bidding at 0... |
| 01:00.40 | [TK]D-Fender | trying shoing a complete call. Certificate setups, peer setups, etc. |
| 01:00.51 | [TK]D-Fender | You know the things that actually constitute your configuration. |
| 01:01.01 | johndropper | my config is fine |
| 01:01.10 | johndropper | thanks for trying help. Im moving on |
| 01:01.13 | johndropper | regards |
| 01:01.35 | [TK]D-Fender | yup... another one of those... |
| 01:01.39 | johndropper | I can get better results with hard knocks than argueing with you |
| 01:01.53 | johndropper | it has been real tho |
| 01:01.55 | [TK]D-Fender | You offer nothing |
| 01:01.59 | johndropper | thanks bro |
| 01:02.08 | [TK]D-Fender | Resulats are likely to remain proportionate |
| 01:02.15 | johndropper | lol ill get it |
| 01:02.24 | [TK]D-Fender | Hopefully so |
| 01:02.30 | johndropper | thanks |
| 01:02.36 | WIMPy | johndropper: None of us have magic glass spheres. Nor as far as I know. |
| 01:02.50 | johndropper | its cool |
| 01:02.52 | [TK]D-Fender | I do, but they are mostly good for burning ants... |
| 01:03.12 | WIMPy | That's not magic, just optics. |
| 01:03.21 | [TK]D-Fender | And when I look inside it looks like it's snowing! |
| 01:03.37 | [TK]D-Fender | It is sufficiently advanced... it seems like magic! |
| 01:04.21 | lanning | WIMPy: would acrylic work for the sphere? :) |
| 01:05.04 | WIMPy | According to the legend it is galss. But we don't know for sure as noone has found a working version, yet. |
| 01:08.23 | jameswf | http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.9.0 there is always this |
| 01:10.17 | [TK]D-Fender | jameswf: Well there shouldn't be a functionaly change that would take this out at that poitn... so what might be suspected as a bug would be on the tracker... not in the changelog... otherwise it'd already be fixed and we wouldn't be hearing about this now. |
| 01:10.24 | [TK]D-Fender | ... Just sayin' |
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| 01:48.43 | RaMcHiP | Woooohoo, got click to dial working |
| 01:48.57 | RaMcHiP | Now I have only have 1 more question, when I dial into my PBX I dont hear any ringing |
| 01:49.19 | RaMcHiP | how do I make incoming calls ring while its connecting? |
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| 01:57.14 | youjelly | I call resetcdr() just before I dial a number, but in the cdr its including the time when the phone is still ringing, can that be catered for? |
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| 03:34.32 | johndropper | vm123 |
| 03:34.57 | johndropper | can anyone see me? |
| 03:35.29 | johndropper | how can I get asterisk to default to ftp even if sftp is offered? |
| 03:35.41 | johndropper | oops rtp and srtp |
| 03:37.04 | johndropper | anyone here |
| 03:37.06 | johndropper | ? |
| 03:40.49 | lvlinux | just dont enable srtp |
| 03:43.05 | lvlinux | and it wont use it |
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| 04:10.21 | johndropper | its enabled i guess |
| 04:10.27 | johndropper | how do I disable? |
| 04:10.32 | johndropper | lvlinux |
| 04:10.45 | johndropper | lvlinux: how do I disable |
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| 04:51.12 | lvlinux | johndropper: gtg but i b back tomorrow---ck google should b easy to find. |
| 04:51.27 | johndropper | lol not |
| 04:52.03 | lvlinux | disable tcp in ur sip.conf for starter---that might take care of it easy. |
| 04:52.08 | lvlinux | b back tomoorow |
| 04:52.11 | lvlinux | goes to bed |
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| 08:24.29 | fidividi | whats the best way to shorten the keep-alive and refresh period for peers/trunks/extensions in asterisk, for those devices/peers that go unreachable/lagged? |
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| 08:45.53 | wdoekes | fidividi: qualifyfreq? |
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| 09:19.07 | bstarek | hello all |
| 09:19.28 | bstarek | I have this very weird issue where only conference call works over nat, but not the rest :) |
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| 09:36.46 | bstarek | anyone? |
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| 10:13.58 | peetaur | in Asterisk 11.9.0, how do I get the "console dial" command? |
| 10:16.47 | peetaur | I have chan_oss built and I remarked out the "noload chan_oss.so" in modules.conf and restarted asterisk, but it doesn't show the dial command in help |
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| 11:02.28 | mirela666 | HI, when doing the SIP/2.0 181 Call is being forwarded I get the: pp_dial.c:958 do_forward: Not accepting call completion offers from call-forward recipient |
| 11:03.07 | mirela666 | is there a way to emove this notice? is some sip header incomplete? |
| 11:04.42 | mirela666 | remove* |
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| 12:31.48 | marceloamorim | guys I had an issue on monday and I did today again, but I couldn`t find the problem yet, the problem is on my asterisk asterisk 11.6-cert2 |
| 12:33.59 | marceloamorim | I have an asterisk in one place and I have tunnel to the local where I have the phones, and run like if was local network, but the phones are being disconected every day on the energy supply |
| 12:34.54 | marceloamorim | every day on the morning they turn on the phones, but the phones can`t registry on the pbx |
| 12:35.17 | marceloamorim | so I need to do "sip reload" then all phones registry again |
| 12:35.58 | marceloamorim | I look to the log on /var/log/asterisk/full and doesn`t have any log for the phones |
| 12:39.31 | marceloamorim | so I wondering when this problem happen again, because it happened on monday and today, if I should get the debug for this or if I try tcpdump on the ethernet |
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| 13:22.32 | saint_ | hi all - is there a way to have some kind of heart bit with asterisk, so when the sip trunk goes down, it attempts to reconnect automatically, instead of having people telling me that "the phone system is not working", and restart asterisk manually ..? |
| 13:22.34 | djhenrya | hi |
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| 13:23.08 | [TK]D-Fender | saint_: Depends on your definition of "goes down" |
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| 13:23.33 | djhenrya | How can I get an event/message/notification from asterisk whenever an incoming call was handled? |
| 13:23.56 | [TK]D-Fender | djhenrya: define "handled" |
| 13:24.02 | saint_ | well.. it happened yesterday, and i did not really have time to troubleshot. All I noticed was that internal calls were working "over the internet". I have digium phones spreaded behind private networks, and it works. The only thing that did not work was incoming / outgoing through the sip trunk. |
| 13:24.30 | djhenrya | inet <-> asterisk <-> me |
| 13:24.43 | [TK]D-Fender | saint_: Lack of details doesn't help in identifying what if anything can be done |
| 13:24.45 | djhenrya | when someone calls me I want to be notified |
| 13:24.52 | [TK]D-Fender | djhenrya: How? |
| 13:25.09 | djhenrya | how what? |
| 13:25.25 | [TK]D-Fender | djhenrya: HOW do you want to be notified? |
| 13:25.29 | saint_ | [TK]D-Fender I understand and agree with that. but is there something in general that people use.. like a script, an add-on, something else, that will prompt asterisk to try to restart itself if it can't connect to the sip carrier ? |
| 13:25.47 | [TK]D-Fender | saint_: Restarting * isn't the problem. |
| 13:25.59 | saint_ | [TK]D-Fender in my case, restarting was the fix.. |
| 13:26.04 | [TK]D-Fender | saint_: your concept of "connection" is flawed |
| 13:26.07 | djhenrya | I would like a message to be sent to me by socker, or http |
| 13:26.12 | djhenrya | socket |
| 13:26.26 | [TK]D-Fender | djhenrya: well it's your dialplan... go call a script before you dial your device |
| 13:26.52 | djhenrya | I searched for an example but couldn't find anything |
| 13:27.14 | djhenrya | I would also like to "click a dial" from my web page... |
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| 13:27.27 | [TK]D-Fender | djhenrya:Well you have to clearly define how you intend to send it... because this is your job to invent |
| 13:27.29 | saint_ | [TK]D-Fender a script for example where asterisk would try to call a number, if it rings back , then it is OK. if there is no ring back, then it tries to restart sip trunk. is this doable ? |
| 13:27.55 | [TK]D-Fender | saint_: Again your definition of what a "connection" is is flawed. |
| 13:28.23 | djhenrya | Fender, look - I want my web application to show who is calling me now |
| 13:28.35 | djhenrya | I also have a sip client to handle the call |
| 13:28.44 | djhenrya | how can I notify the browser about call data? |
| 13:29.00 | [TK]D-Fender | djhenrya: that's for you to code. |
| 13:29.11 | djhenrya | code where? |
| 13:29.33 | [TK]D-Fender | djhenrya: How can someone just throw information at a web browser? Web browsers tend to pull information... not have it thrown at them |
| 13:29.55 | djhenrya | yeah yeah.. let me implement the polling/websocket |
| 13:30.02 | [TK]D-Fender | djhenrya: you make your script on your * server. You have your dialplan call that script before dialing your device |
| 13:30.09 | djhenrya | but what do I poll? I saw asterisk 12 has REST API |
| 13:30.36 | djhenrya | okay, that's one. Can you give me an example for such a script? |
| 13:30.48 | [TK]D-Fender | If you want to set something raw I'd recommend using AMI and a Custom Event |
| 13:31.47 | djhenrya | ok, thanks. I am looking into it |
| 13:31.53 | djhenrya | I have another question |
| 13:32.10 | djhenrya | How can I implement a "click-a-dial" with asterisk? |
| 13:32.19 | [TK]D-Fender | djhenrya: AMI Originate <- |
| 13:33.54 | *** join/#asterisk Quastor (~chatzilla@mail.edpnet.net) |
| 13:34.39 | djhenrya | PERFECT! thanks:) |
| 13:36.01 | Quastor | Hi when we try to send from an hardware fax with an ata controller we receive: "Failed to initialize UDPTL, declining image stream" and the fax is unreadable with the receiver. Any idea |
| 13:36.26 | Katty | hi |
| 13:39.14 | jameswf | Quastor: don't fax over IP |
| 13:40.00 | jameswf | Katty: Colorodo and Washington aren't awake yet so no one is high |
| 13:41.41 | Katty | hugs jameswf |
| 13:44.05 | Quastor | jameswf: well if possible I didn't connect any analog device on asterisk, but ye |
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| 14:03.55 | MaliutaLap | jameswf: it'll be a while until they wake up, then they'll have to deal with the munchies ;) |
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| 14:08.07 | *** join/#asterisk XandriX (xandrix@gateway/shell/anapnea.net/x-enwvpiwwvgrzbvwc) |
| 14:08.42 | XandriX | i know this is not the proper place to ask but anyone here ever mess around with avaya phone systems ? |
| 14:10.00 | MaliutaLap | XandriX: no, I can't convince people to lend me the kit to play with |
| 14:10.14 | XandriX | MaliutaLap: okies |
| 14:11.01 | jameswf | XandriX: back when they were toshiba |
| 14:11.20 | XandriX | jameswf: o.O |
| 14:11.34 | XandriX | our model used to be nortel not toshiba you must be old :P |
| 14:11.47 | jameswf | late 90s |
| 14:11.55 | jameswf | early 00s |
| 14:12.33 | jameswf | I have worked on merlins with 5.25 floppy's.... I have seen things |
| 14:13.37 | MaliutaLap | jameswf: things you can never unsee ... we've all been there :) |
| 14:15.53 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-xdtlxqfpylwwqptp) |
| 14:15.53 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:17.58 | XandriX | jameswf: no offense was intended btw :) |
| 14:19.57 | jameswf | XandriX: part of my job description is punching bag, I don't offend easy.... |
| 14:22.06 | XandriX | i hear ya |
| 14:22.22 | XandriX | im stuck with a bcm450 at work and i need to find a way to actualy access root without mounting the hdd in another machine |
| 14:22.27 | XandriX | so far quite frustrating xD |
| 14:23.19 | MaliutaLap | would mounting the drive in another system be _that_ bad ;) |
| 14:38.32 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:38.41 | XandriX | MaliutaLap: yeah it means down time cuz its our only phone system and they dont like overtime here but we must fix things and do as they ask xD |
| 14:38.43 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:39.30 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-vfzrbkhavuwangjh) |
| 14:39.55 | MaliutaLap | XandriX: people need to learn they can't have it both ways ... do you want it fixed? or do you want to avoid paying overtime/time-in-lieu |
| 14:40.32 | MaliutaLap | mostly my contracts have time-in-lieu ... so they really don't mind, mainly because they have no intention of giving me the time back anyhow :( |
| 14:41.25 | XandriX | yeah |
| 14:41.28 | XandriX | ill figure something out |
| 14:41.54 | MaliutaLap | I suggest threatening the disk with a super magnet ;) |
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| 14:45.51 | alami | i want to secure an asterisk server that i'm using only for conference, i want only that Active directory or people that have smartcard can start conference |
| 14:46.01 | alami | is there a way to secure this server? |
| 14:46.15 | jameswf | XandriX: why not reboot in to single user |
| 14:46.34 | jameswf | alami: AGI |
| 14:49.44 | fidividi | wdoekes: thanks a lot for pointing out "qualifyfreq" to me. Do they go on PEER settings or is there a way to generalize it on sip.conf? I am interested in the latter. |
| 14:50.49 | [TK]D-Fender | alami: How do you start a conference at all? |
| 14:51.37 | *** join/#asterisk boom^time (~boom^time@75-151-20-174-Michigan.hfc.comcastbusiness.net) |
| 14:51.42 | XandriX | jameswf: still gonna kill our only current running phone system xD |
| 14:51.58 | *** join/#asterisk LemensTS (~t15@8.33.19.98) |
| 14:52.05 | jameswf | XandriX: yes but for 5 minutes rather than 30 |
| 14:52.11 | XandriX | jameswf: maybe one night when there is overtime ill sneek into that room with a usb to serial adapter and the proper cable for it |
| 14:52.31 | XandriX | jameswf: that bcm450 takes about 45 minutes for everything to be back and fully running its a piece of crap xD |
| 14:52.37 | LemensTS | Anyone know why asterisk does not announce the 3rd digit of parked lot extension? http://pastebin.com/qdf0ss5w |
| 14:52.59 | alami | [TK]D-Fender if a call start with the number 5, for example 5555 it will ask for a password , and the make a conference |
| 14:53.27 | [TK]D-Fender | LemensTS: I'd be wondering if they were waiting long enough beofre finalizing the attended transfer or not. |
| 14:53.45 | [TK]D-Fender | alami: That's up to you and your dialplan. |
| 14:53.50 | alami | [TK]D-Fender : my Problem i want to make this conferencebridge connect to the pstn, so that i can call from home and from outside |
| 14:54.07 | [TK]D-Fender | alami: That idea is backwarks. |
| 14:54.20 | alami | [TK]D-Fender: and i don't want that everybody can dial and start a conference |
| 14:54.25 | XandriX | jameswf: also there seems to be a private key for the root user on the file system if i could get my hands on it maybe i could do something |
| 14:54.29 | [TK]D-Fender | alami: It doesn't connect to the PSTN ... your PSTN connection leads to that dialplan |
| 14:54.39 | LemensTS | TKD-Fender: I call in to sip phone, wait 5 seconds after I answer, I press park and listen to it say 70 and then it hangs up. CLI also shows it does not play the 1 |
| 14:54.50 | XandriX | cuz for somereason they give you access to a logfile that is essentialy ls -al / but that traverses all directory |
| 14:54.53 | LemensTS | Polycom 321 btw |
| 14:55.00 | [TK]D-Fender | LemensTS: I'd elaborate on this "press park" |
| 14:55.12 | jameswf | 70 is the lot 701 is the pickup |
| 14:55.13 | [TK]D-Fender | LemensTS: and I'd try doing it the NORMAL way for comparison |
| 14:55.21 | jameswf | ~Freepbx |
| 14:55.21 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 14:55.24 | jameswf | btw |
| 14:55.59 | alami | [TK]D-Fender: but at the moment any body can call and start conference, and i want to limite that |
| 14:56.11 | jameswf | alami: AGI |
| 14:56.41 | alami | jameswf: can i make a use of smartcard authentication with AGI? |
| 14:57.07 | [TK]D-Fender | alami: Again, it's your dialplan.. if you don't want 1 place to be able to start it.... don't put the apps there to do it |
| 14:57.13 | LemensTS | Thats odd. I press transfer then 700 and it says 701...its probably my Call Park code in polycom cfg: efk.efklist.1.action.string="$FTransfer$$FDialpad7$$FDialpad0$$FDialpad0$$FDialpadPound$$Cp3$$Chu$" |
| 14:57.16 | jameswf | alami: you can make anything with agi |
| 14:57.57 | jameswf | alami: I have only been skimming but I think you want DisA |
| 14:58.14 | [TK]D-Fender | LemensTS: Cp3 = "pause 3 seconds" by any chance? Perhaps not quite long enough? |
| 14:58.26 | [TK]D-Fender | jameswf: He doesn't |
| 14:58.52 | [TK]D-Fender | jameswf: just authing a caller I wouldn't pump through a secondary channel like that. |
| 14:59.21 | [TK]D-Fender | jameswf: And it dies fast on timeout.... |
| 14:59.39 | jameswf | [TK]D-Fender: I thought I saw he was passing calls through his phone system by calling in |
| 14:59.58 | [TK]D-Fender | jameswf: We're only talking about a confbridge here... |
| 15:00.27 | jameswf | [TK]D-Fender: I thougt he was doing some voodoo to use a confbridge as disa |
| 15:00.41 | jameswf | [TK]D-Fender: stranger things have happened |
| 15:00.59 | [TK]D-Fender | jameswf: jameswf No, he doesn't understand the concept of separating your dialplan in contexts and actually configuring dilaplan flow to demand auth where you want it, etc. |
| 15:02.05 | *** join/#asterisk BakaKuna (~Thunderbi@office.voys.nl) |
| 15:02.17 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
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| 15:04.06 | leifmadsen | fuggit! auth all the calls! |
| 15:04.34 | XandriX | jameswf: apparently i haveto either hookup a laptop with a crossover cable to access it properly on the OAM port or use a serial cable hehe |
| 15:05.55 | jameswf | leifmadsen: minutes are cheap |
| 15:06.03 | leifmadsen | ikr? |
| 15:06.16 | *** join/#asterisk bmurt (~brendan@74-95-169-22-Philadelphia.hfc.comcastbusiness.net) |
| 15:07.06 | jameswf | I feel old... I had to stare at that for like 45 seconds |
| 15:07.16 | leifmadsen | :) |
| 15:07.26 | leifmadsen | crazy kids and your acronyms |
| 15:07.28 | leifmadsen | get off my lawn! |
| 15:08.07 | jameswf | I was the good parent reading my kids text messages.... After a few weeks of that I couldn't afford to lose any more IQ points so I stopped |
| 15:09.07 | LemensTS | That did the trick TK-FENDER, thanks |
| 15:11.40 | leifmadsen | jameswf: lolz |
| 15:12.03 | djhenrya | lol. |
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| 15:15.00 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 15:21.14 | [TK]D-Fender | LemensTS: You're welcome |
| 15:21.53 | wdoekes | fidividi: you could just test it in general and see if it works |
| 15:23.15 | wdoekes | spoiler: the option is parsed in the global reload_config() |
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| 15:33.10 | alami | [TK]D-Fender: can i make only user with Active directory account can start a conference in my dialplan? |
| 15:34.05 | *** join/#asterisk raspberrypifan (~textual@71-22-220-224.gar.clearwire-wmx.net) |
| 15:39.07 | [TK]D-Fender | alami: Asterisk knows nothing of "active directory". Any lookups you do is in a script of your making |
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| 15:41.09 | alami | [TK]D-Fender: that's why i was asking because you have sure more experience then me, and my question is quite simple |
| 15:41.35 | jameswf | alami: google "<Linux scripting language of choice> Active Directory" then google "asterisk AGI" |
| 15:41.38 | alami | and i want only to authorize some ppl to call the conference bridge and start a conference |
| 15:41.46 | [TK]D-Fender | alami: I have no experience with Active Directory. I just know that Asterisk has NO functionality for talking or caring about it, so it's up to you to code. |
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| 15:42.17 | alami | ok allright, thanks a lot |
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| 15:57.08 | MaliutaLap | [TK]D-Fender: all you need to know about MS-AD is that it's a borked LDAP implementation :) |
| 16:02.11 | jameswf | MaliutaLap: all you need to know about MS-AD is that it's a Microsoft implementation FTFY |
| 16:04.25 | MaliutaLap | jameswf: you're right any MS implementation is borked - I shouldn't have been redundant in my use of language. It's just that I'm used to building redundancy into any system I can ;) |
| 16:04.54 | johndropper | tk at it again |
| 16:08.13 | boom^time | Hey is there a good way in an asterisk dial plan to be able to dial an extension during a call to xfer the person you're talking to to an extension and places the operator on hold until they are finished and then brings them back together? |
| 16:10.44 | jameswf | boom^time: why would you want the operator to hold |
| 16:11.17 | boom^time | jameswf, I suppose they could follow along and be muted. |
| 16:11.21 | *** join/#asterisk LemensTS (~t15@8.33.19.98) |
| 16:11.47 | jameswf | boom^time: couldn't you set the hangup destination back to the operator so they could move on |
| 16:12.43 | LemensTS | Connected Line ID: 12129991111 Eff. Connected Line ID: 12129991111 if i do show core channel channelname during a call it shows these 2 stats. Can I get those via the dialplan? |
| 16:12.45 | boom^time | the operator is part of a queue from a different system accessible by a single number |
| 16:13.05 | boom^time | no guarantee it'l get back that operator, who ever is next in line will get it |
| 16:15.41 | marceloamorim | guys, can I make my asterisk insecure for one ip? like I have an interface gsm and this interface to send me the callerid I need to receive this call "without registration" but now I have this message "Sending fake auth rejection for device 03799841120 <sip:03799841120@192.168.2.254:5060;user=phone>;tag=c17f3a26f3234e877d1ab71c72c1ad8c" |
| 16:15.54 | marceloamorim | but I need this just for this 192.168.2.254 |
| 16:16.01 | marceloamorim | is it possible? |
| 16:16.53 | jameswf | boom^time: I guess I don't understand the call flow or locking 2 people up. It can be done but your best bet is probably a conference call unless you keep a table of "busy agents" once the call transfers out the agent is technically availible |
| 16:20.14 | *** join/#asterisk ghoti (~paul@205.233.216.25) |
| 16:21.07 | boom^time | jameswf, I see. What method do you use to have asterisk allow a transfer or conference during a call? |
| 16:22.37 | jameswf | boom^time: google "asterisk features.conf" |
| 16:22.53 | boom^time | jameswf, thank you |
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| 16:26.09 | marceloamorim | I set alwaysauthreject=no and allowguest=yes, but I wish to open just for this device |
| 16:31.35 | *** join/#asterisk Mango45 (~Mango45@d209-89-208-218.abhsia.telus.net) |
| 16:32.26 | Mango45 | Using the dial plan, is it possible to find a list of codecs that a peer wants to use? I can see it in the invite if I look at sip debug, but not sure how to use it progmatically. |
| 16:33.05 | jameswf | Mango45: explain "codes" |
| 16:33.18 | Mango45 | like g.711, g.722, g.729, gsm, and so on. |
| 16:34.21 | jameswf | Mango45: this is handled by the protocol. You say what you are willing to use. They say what they are willing to use and the protocol negotiates |
| 16:34.35 | jameswf | 1st match wins I believe |
| 16:34.50 | Mango45 | The purpose of the exercise is to disable transcoding. |
| 16:35.16 | Mango45 | If one peer allows G.711, and the other allows G.722, G.711, Asterisk will transcode instead of using the first common. |
| 16:35.18 | jameswf | MaliutaLap: then only offer G.711 |
| 16:35.34 | Mango45 | I want to use G.722, if both peers support it. |
| 16:35.59 | Mango45 | I know I can set the SIP_CODEC variable, but not sure how to determine what codecs are available for an incoming call. |
| 16:36.16 | jameswf | so dissalow all, allow g722, g711 |
| 16:36.29 | jameswf | order matters |
| 16:36.40 | Mango45 | Yep. |
| 16:36.53 | jameswf | then it is in the hands of the sip gods |
| 16:37.17 | Mango45 | I want it to be in my hands ;) |
| 16:37.22 | Mango45 | I disagree with the sip gods. |
| 16:37.58 | jameswf | MaliutaLap: write your own sip stack |
| 16:38.28 | Mango45 | Sure, that'll only take me 15 or 20 minutes. :) |
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| 16:40.51 | jameswf | Mango45: maybe you should learn to accept the decisions of the sip gods :) |
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| 16:41.48 | jameswf | Mango45: to get a clear understanding of everything going on do a tcpdump on a call |
| 16:43.16 | fidividi | slaps fidividi around a bit with a large trout |
| 16:44.21 | fidividi | wdoekes: how can i verify once the settings are in place? |
| 16:44.34 | jameswf | fidividi: no paraphyletic abuse |
| 16:45.20 | fidividi | jameswf: im new to all this, i was seeing how to put usernames in beginning instead of manually writing it.. |
| 16:48.10 | fidividi | jameswf: apparently it is all done manually, right? |
| 16:49.26 | marceloamorim | anyone knows if its possible? |
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| 17:08.08 | otubo | Hello guys, I have a question: If I use a chan_dongle with a GSM modem with voice feature enabled, can I simply call the modem and redirect it to an extension? Is it possible? |
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| 17:51.02 | igcewieling | I'm getting the following error in the CLI, if it was a NAT or firewall issue I'd expect the error to be dest unreachable or something like that. Using Asterisk 11. Error: "[May 14 13:50:29] WARNING[2408]: chan_sip.c:3907 __sip_xmit: sip_xmit of 0x7f7e901987d0 (len 610) to 209.220.119.87:50663 returned -2: Interrupted system call" |
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| 18:13.14 | marceloamorim | otubo: I've never used the chan_dongle, but I did a research some days ago and yes, you can do this |
| 18:14.19 | marceloamorim | this page help me with this dongle thing, www.asteriskbrasil.org.br/forum/topic/59-instalando-e-utilizando-chan-dongle/ |
| 18:15.16 | marceloamorim | it is in portuguese, but the samples are in english, so I think you may understand |
| 18:15.47 | marceloamorim | look for [dongle-incoming] |
| 18:16.46 | *** join/#asterisk danitetus (~danitetus@90.Red-83-45-47.dynamicIP.rima-tde.net) |
| 18:17.12 | igcewieling | very very few people use chan_dongle so you won't find many people who can help |
| 18:17.57 | otubo | marceloamorim, na verdade eu sou brasileiro :) então documentação em pt_BR é OK :) Thanks a lot! |
| 18:18.59 | marceloamorim | você cometeu uns erros em ingles que eu geralmente cometo tambem, mas não esperava que fosse brasileiro =) mas tudo bem =) |
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| 18:58.11 | navaismo | The ${MACRO_EXTEN} should be available at the beginning of any MACRO or i need to set up first somewhere? |
| 18:58.30 | navaismo | a NoOp(${MACRO_EXTEN}) show nothing :( |
| 18:58.54 | [TK]D-Fender | Show us |
| 19:01.37 | navaismo | http://pastebin.com/ganav08h |
| 19:03.34 | [TK]D-Fender | -- Executing [90455530501800@from-service-level:1] Gosub("SIP/6001-00000015", "permiso,~~s~~,1(7)") in new stack <- do you see the word "macro" in there? |
| 19:05.40 | navaismo | hmm |
| 19:07.56 | *** join/#asterisk timahvo1 (~rogue@197.237.174.64) |
| 19:08.16 | navaismo | but no gosub in the dialplan, this is a AEL dp |
| 19:08.55 | [TK]D-Fender | AEL gets parsed back to standard dialplan. |
| 19:09.02 | [TK]D-Fender | And you didn't look at what that was |
| 19:10.53 | navaismo | ok so the line _9045ZXXXXXXXXX => &permiso(7); is tranated to Gosub(permiso,~~s~~,1(7)), but the wiki has as an example the ¯o-name to jump at |
| 19:11.09 | [TK]D-Fender | Where? |
| 19:11.11 | navaismo | https://wiki.asterisk.org/wiki/display/AST/AEL+Macros |
| 19:11.50 | [TK]D-Fender | See that giant yellow block in the middle of the page? |
| 19:12.06 | navaismo | OMG |
| 19:12.49 | navaismo | actually in my pc is light-yellow but how in the hell i pass that information with that bold and warning icon |
| 19:12.51 | navaismo | jesus |
| 19:13.13 | navaismo | sometimes i say to myself(what...) i need to change profession |
| 19:14.15 | navaismo | starting the migration to normal dialplan |
| 19:18.31 | navaismo | Thank you for open my eyes |
| 19:19.30 | marceloamorim | [TK]D-Fender: do you know how can I fix the sending fake auth rejection for device? |
| 19:19.50 | [TK]D-Fender | Don't auth your peer |
| 19:20.36 | *** join/#asterisk Corey84 (~Corey84@unaffiliated/corey84) |
| 19:20.41 | marceloamorim | I don`t want auth this peer |
| 19:21.25 | marceloamorim | because I don`t know why when I use sip peer the callerid didn`t repass to my asterisk |
| 19:21.35 | marceloamorim | it is a gsm gateway |
| 19:21.45 | [TK]D-Fender | You configured your peer wrong |
| 19:22.02 | [TK]D-Fender | insecure=port,invite <------------------------- |
| 19:22.52 | marceloamorim | I should set this insecure for this peer? |
| 19:23.20 | [TK]D-Fender | Clearly |
| 19:23.41 | marceloamorim | ty I´ll try again |
| 19:40.58 | navaismo | Done! [TK]D-Fender was easier add another argument to the macro and to the gosub lol |
| 19:41.19 | [TK]D-Fender | usually is.... |
| 19:43.14 | [TK]D-Fender | ${MACRO-EXTEN} is practically worthless crap since it's always best to tell your macros what they need to know vs trusting that where you call it from is always relevant. |
| 19:43.27 | marceloamorim | omg, I found a problem that give me the problem for this callerid thing |
| 19:43.44 | marceloamorim | I just remove from sip.conf the callerid line o.O |
| 19:44.51 | [TK]D-Fender | Forcing a callerID on your inbound peer... would be a horrible mistake as you'd be overriding whatever the device sends. |
| 19:45.52 | marceloamorim | yeah, I didn't know that, how bad I'm, I shouldn't try to be like autodidact, I don't have skills for that |
| 19:46.28 | marceloamorim | asterisk will give me white hair |
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| 20:56.30 | fidividi | guys I am using XenServer and my voip box is a VM in there, by networking is done through routed-networks internally, but my VM has public IP. My peers go unreachable/lagged/... Can anyone help me figure this out? |
| 20:58.48 | fidividi | I tried playing around with "qualifyfreq" and "qualify" and "keepalive", but i just don't know what is causing it to fix it... I appreciate any help. |
| 21:00.21 | woleium | lo all :) |
| 21:00.44 | woleium | In device and user mode how do i allow a handset to makecalls if it's not logged in? |
| 21:00.55 | woleium | I'm getting "your concurrent call limit has been reached" :( |
| 21:01.56 | woleium | oops, wrong channel |
| 21:02.00 | marceloamorim | you can look if the phones need to pass through the internet to get your ip of pbx |
| 21:02.13 | woleium | this is asterisk, I meant freepbx... |
| 21:04.30 | fidividi | marceloamorim: everything is on internet, all phones/peers/pbx are all on internet |
| 21:07.18 | marceloamorim | you can try ping and traceroute to see if the problem is on the internet, and try to identify if there is a "bottleneck" ( sorry I don't know if this word mean what I want to say) |
| 21:09.38 | marceloamorim | when I use the internet to register a phone, in Brazil I set qualify to 10000 |
| 21:10.04 | marceloamorim | sometimes I use 25000, because the bottleneck is huge |
| 21:10.17 | fidividi | marceloamorim:And the only thing you set is qualify? do they ever go UNREACHABLE or LAGGED? |
| 21:11.08 | marceloamorim | when the hit above this set up, 10000 or 25000 |
| 21:11.17 | fidividi | marceloamorim: What do you have for NAT? yes, no, route, never? |
| 21:11.21 | marceloamorim | but 25000 is just when the network stopped |
| 21:12.21 | fidividi | marceloamorim: And do you use keepalive or qualifyfreq or anything else? |
| 21:12.42 | marceloamorim | I think the parameters was changed for no, force_rport, yes and comedia |
| 21:12.49 | marceloamorim | but I don't have sure |
| 21:13.26 | marceloamorim | but I'm not sure** |
| 21:14.56 | *** join/#asterisk jonno11 (~jonno11@amigopod.rave.ac.uk) |
| 21:15.19 | fidividi | marceloamorim: for instance, from my home IP to PBX IP, if I ping, I get 40ms, but on asterisk, "sip show peers" shows for my peer something like 200~300ms, and sometimes it changes to 1000ms even.. which is not normal, because i tried pining the time when it is showing 1000ms, and my ping is still 40ms |
| 21:16.01 | *** join/#asterisk RaMcHiP (~RaMcHiP@wsip-98-190-149-169.ph.ph.cox.net) |
| 21:16.13 | fidividi | marceloamorim: all my accounts have good internet connections, so it doesn't make sense at all... |
| 21:16.50 | marceloamorim | this is normal to have ping like 40ms and peer 200~300, I have ping like 1ms on localnet and I have peer with 74ms |
| 21:17.09 | RaMcHiP | Thanks to all that helped me |
| 21:17.14 | RaMcHiP | I have Click to dial working through Vtiger |
| 21:17.22 | RaMcHiP | Much appreciated! |
| 21:17.35 | marceloamorim | but I don't know why =( sorry |
| 21:18.21 | fidividi | marceloamorim: thanks anyway, do you use keepalive or qualifyfreq or anything else except qualify? |
| 21:19.37 | marceloamorim | I used this qualify=25000 because I want to check the device, but I set qualify=no when this makes me have headache |
| 21:19.58 | marceloamorim | I'm not sure what is the better option when you have qualify |
| 21:20.25 | marceloamorim | maybe another person can help us on that |
| 21:20.51 | TechSmurf | wifi phone or android/ios device with app? |
| 21:24.27 | fidividi | TechSmurf everything really, corded ip phones like snom or escene or atcom, ios device with bria, android with bria |
| 21:24.51 | fidividi | TechSmurf: sometimes (rarely) even my provider trunks go unreachable |
| 21:24.59 | TechSmurf | sorry, that was a question |
| 21:25.05 | TechSmurf | not an attempt to help |
| 21:25.55 | fidividi | TechSmurf: ok, my bad, sorry.... definitely device with app... |
| 21:26.23 | *** join/#asterisk aness (~aness@2a02:fe0:c310:8b20:e5ab:b5cb:fecd:77c9) |
| 21:28.57 | RaMcHiP | Now that I have all of this working what kind of fancy stuff can I do? |
| 21:29.27 | RaMcHiP | Does anyone know how to make it ring on the callers side before we answer the phone? Its just dead silent and I would like a standard ringing sound. |
| 21:31.55 | marceloamorim | I don't know if the indications.conf help you on this RaMcHiP |
| 21:33.32 | RaMcHiP | It cannot hurt to look! |
| 21:33.47 | RaMcHiP | I am loving this, having the most fun I have had in years playing around with elastix |
| 21:33.56 | RaMcHiP | Thanks! |
| 21:42.12 | igcewieling | I use qualify=10000 and qualifyfreq=120 |
| 21:45.57 | *** join/#asterisk Ice_Strike (~Ice_Black@84.92.51.164) |
| 21:46.11 | Ice_Strike | Anyone used nagios to monitor asterisk? |
| 21:48.26 | navaismo | o/ icinga a long time ago |
| 21:48.53 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 21:54.51 | fidividi | igcewieling: Do u ever get lagged/unreachable devices? |
| 21:57.01 | fidividi | igcewieling: Have you customized anything else? Like: registertimeout registerattempts minexpiry maxexpiry defaultexpiry ? |
| 21:57.15 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 22:00.16 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 22:03.48 | *** join/#asterisk Corey84 (~Corey84@unaffiliated/corey84) |
| 22:04.56 | *** join/#asterisk bkruse (~Adium@74.51.115.113) |
| 22:12.43 | RaMcHiP | I looked in the indicationss.conf and couldnt really decypher it :) It was saying that all of those variables should be set through the UI |
| 22:12.59 | RaMcHiP | is there a setting on my inbound trunk that will ring while it connecting? |
| 22:13.13 | RaMcHiP | When I get incoming calls the caller does not hear ringing or anything |
| 22:13.44 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 22:19.32 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 22:22.11 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
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| 22:38.35 | *** join/#asterisk aness (~aness@2a02:fe0:c310:57f0:38bd:6cdf:3a54:ad9) |
| 22:43.39 | RaMcHiP | Perfect, got ringing working now. |
| 22:43.56 | RaMcHiP | So what fun stuff can I play around with now? |
| 22:43.58 | RaMcHiP | :) |
| 22:51.37 | *** join/#asterisk bmurt (~brendan@64-121-18-195.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 22:53.34 | *** join/#asterisk swiss__ (~swiss__@70.184.40.66) |
| 22:55.05 | swiss__ | hey all, how can i get a macro to hang up and or stop running the dialplan? |
| 22:56.18 | swiss__ | dialplan and macro here: http://pastebin.com/JZERTFVJ |
| 23:01.42 | *** join/#asterisk insha (~insha@mobile-166-137-185-188.mycingular.net) |
| 23:05.23 | *** join/#asterisk dfighter (~someone@arcemu/staff/dfighter) |
| 23:05.37 | navaismo | set the macro result to abort |
| 23:05.44 | *** join/#asterisk edong23 (~quassel@mptc-dhcp-50-220.mptelco.com) |
| 23:05.59 | navaismo | ABORT: Hang up both legs |
| 23:06.04 | swiss__ | ah |
| 23:06.16 | swiss__ | i was searching for hangup both legs but coudnt find that |
| 23:06.19 | swiss__ | than you good sir |
| 23:17.02 | *** join/#asterisk ralphmazio (~ralphmazi@cpe-066-057-255-154.nc.res.rr.com) |
| 23:17.28 | ralphmazio | How do you enable srtp in asterisk. I'm not concerned with tls. |
| 23:19.23 | navaismo | encryption=yes |
| 23:19.34 | navaismo | in the sip peer settings |
| 23:24.46 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:57.15 | igcewieling | fidividi: I seldom get lagged or unreachable devices unless there is an actual problem. I do not normally change the registry and expiry settings. For devices behind NAT I usually set the qualifyfreq to 30 |
| 23:57.54 | fidividi | igcewieling: what about keepalive |
| 23:57.55 | fidividi | ? |
| 23:58.03 | igcewieling | fidividi: never used it |
| 23:58.39 | igcewieling | we have a somewhat unique setup where almost all sites are on dedicated IPs |
| 23:59.04 | fidividi | then you don't really have NATing isses |
| 23:59.08 | fidividi | issues* |
| 23:59.32 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 23:59.51 | igcewieling | That depends. The majority of our sites have no NAT for voice, but we have a number (about 20 or so) sites which use NAT |