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| 00:25.00 | jeffspeff | is this still the recommended way to run * as non-root? http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-13-SECT-4.html |
| 00:30.14 | [TK]D-Fender | 3rd ed is still free too... |
| 00:30.16 | [TK]D-Fender | ~book |
| 00:30.17 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 00:30.53 | [TK]D-Fender | And the book isn't wrong in what it describes |
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| 04:49.34 | snadge | exten => _X.,n,Set(x-Privacy=${IF($["${SIP_HEADER(Privacy)}"="id"]?${SIP_HEADER(Privacy)}:off)}) |
| 04:49.54 | snadge | just so i've got this right.. this sets the header Privacy to off .. if it is equal to id ? |
| 04:50.22 | snadge | hmm.. i mean.. it sets x-Privacy to.. ergh.. i have no idea |
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| 08:45.17 | ruied | before dial() I want to check if the number of ${EXTEN} is inside of a file ExternDID.txt . what is the best method so * can search if there is a matching ARG1 inside the file ? |
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| 08:52.52 | ChannelZ | I'd write an AGI to do it |
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| 08:54.37 | joj[] | Can anybody hint me what module the dialplan function FILE is part of? I |
| 08:55.27 | ruied | ChannelZ, ok |
| 08:57.14 | joj[] | Nevermind; it's func_env. |
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| 11:58.29 | slima | Hm, I have sip trunk asterisk <> SONUS SBC1000 And it was working well. On my site I dont made any changes, but now I see via SIP debug only 404: http://pastebin.com/3S9fiNc1 |
| 11:59.46 | slima | What could it be? |
| 12:03.39 | file | that is an OPTIONS request, not a call or anything, and it is looking for the extension 's' in context 'incoming', if it doesn't exist it will send back a 404 Not Found |
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| 12:20.08 | slima | file: yeah, but why it is looking for extension? |
| 12:20.34 | file | because an OPTIONS request is to be treated as if it were an INVITE, except it does not establish a call |
| 12:20.52 | file | so the resulting response should be if an INVITE came in |
| 12:21.01 | file | thus the presence of an extension is taken into account |
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| 12:28.43 | slima | file: I don't understand you in 100%. But, this extension 's' must exist for register? |
| 12:29.03 | file | no |
| 12:29.18 | file | the log you provided showed them sending an OPTIONS request, which is just a "hey are you there" basically |
| 12:29.20 | file | that's it |
| 12:29.28 | file | if you have other problems, they aren't in the log |
| 12:30.11 | slima | Hm, this is all debug for this peer. |
| 12:32.55 | Samot | What does your register string look like? |
| 12:33.01 | Samot | Just mask the password. |
| 12:33.42 | slima | I dont know, It is SONUS SBC1000 PBX in my client office. |
| 12:34.22 | Samot | So it's Provider -> SBC -> Asterisk? |
| 12:35.44 | Samot | Also, in that trunk config. |
| 12:36.00 | Samot | You don't need canreinvite=no, specially not twice. |
| 12:36.07 | slima | Asterisk -> SBC -> clients |
| 12:36.09 | Samot | directmedia= replaces it. |
| 12:36.49 | slima | oh, you are right. |
| 12:37.08 | Samot | So that debug.. |
| 12:37.19 | Samot | What is what? |
| 12:37.47 | Samot | The SBC sent the OPTION to Asterisk? |
| 12:38.56 | Samot | Ahh, looks that way. |
| 12:39.08 | slima | yea, 'remote' == IP of SBC |
| 12:39.19 | Samot | So how does the Asterisk and the SBC communicate? |
| 12:39.30 | slima | sip == asterisk |
| 12:39.32 | Samot | PEER or Registration? |
| 12:41.42 | Samot | Because if you are registering to the SBC from Asterisk, your register string should be register> user:pass@host/user |
| 12:42.05 | Samot | Otherwise the SBC probably sees your Contact as s@asteriskip |
| 12:42.11 | Samot | instead of user@asteriskip |
| 12:42.21 | slima | No, I dont register to SBC |
| 12:42.28 | slima | SBC register to Asterisk |
| 12:44.43 | slima | http://imgur.com/a/WboQ9 |
| 12:44.47 | slima | this is config from SBC |
| 12:47.03 | Samot | Well.. |
| 12:47.11 | Samot | Where it says PEER Username... |
| 12:47.20 | Samot | Put the PEER Username |
| 12:47.39 | Samot | Anonymous basically will get you 's' extension. |
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| 13:17.55 | slima | Samot: They change peer/local to Vandemoortele, you see it at debug. |
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| 13:19.37 | guest9039 | Hi, I 've configured tls in manager.conf. How can I even tell that it is working? It is binded to 0.0.0.0:5039 but when I do netstat -peanut | grep 5039 I get nothing. |
| 13:19.41 | guest9039 | Manager reload shows no errors. |
| 13:21.30 | guest9039 | nmap show 5039/tcp closed unknown |
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| 13:27.59 | guest9039 | SSL error loading private key file. </etc/pki/asterisk.pem> |
| 13:28.02 | guest9039 | Ah, I see this. |
| 13:37.18 | jjrh | I got a issue with pjsip where it's not including the port (5060) in the 100 trying |
| 13:37.53 | jjrh | My provider's equipment apparently can't handle it so they keep sending invites |
| 13:38.24 | jjrh | as far as I can tell all my configs include the port - anyone know the right magic I might be forgetting? |
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| 14:28.00 | swimmercol | Hi folks, I am trying to |
| 14:28.42 | swimmercol | asterisk's voicemail with the option to store the password in the secret.conf |
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| 14:28.42 | swimmercol | when i update my pasword i am getting this error: |
| 14:28.42 | swimmercol | [Sep 1 10:27:41] ERROR[5484][C-0000000c]: app_voicemail.c:14056 write_password_to_file: Error writing voicemail password to /var/spool/asteriskvm/voicemail/default/1234/secret.conf |
| 14:29.06 | swimmercol | I tried setting permissions on that folder and doesn't work |
| 14:29.10 | cervajs2 | can i detect call forward from sip phone and put the call to specific context? |
| 14:29.44 | swimmercol | cervajs2: usually call forward comes with Diversion header |
| 14:30.24 | swimmercol | so you can check if that SIP header exists then route the call to the destionation you want |
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| 14:47.49 | mason | Hey all. I've run Asterisk at several sites in the past, but not for a while. The place I used to prefer as a POTS gateway (Junction Networks) doesn't seem to be in the business of providing access to self-hosted Asterisk any more. Is there a trusted source of listings of businesses that sell gateway access? |
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| 14:51.46 | tulga | I'm trying to connect asterisk to Atcom AT530p VoIP. anyone here experience? |
| 14:55.53 | [TK]D-Fender | There are plenty of guides out there for it |
| 14:56.42 | [TK]D-Fender | Like the ones it took me < 10 seconds to find |
| 14:57.00 | [TK]D-Fender | So what step is failing for you in doing this? |
| 14:57.30 | tulga | [TK]D-Fender: yes, I followed SIP configuration guide, but it is not sending any request to asterisk |
| 14:58.13 | tulga | If I use another IP Phone (Grandstream) it is regitered |
| 14:58.33 | [TK]D-Fender | Show us |
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| 15:15.45 | jjrh | from above - I figured it out, didn't have the right stuff in external_media_address, and external_signaling_address |
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| 17:24.35 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 17:25.01 | *** topic/#asterisk by putnopvut -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.11.0 (2016/09/01), 11.23.0 (2016/07/21), Standard: 14.0.0-beta2 (2016/08/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 17:59.22 | jeffspeff | i'm following a tutorial (https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial) for making client certs. one of the options is to specify the client hostname or ip address. what should you do if you don't know the exact hostname or ip that the client will have? can you just leave that blank or something? |
| 18:01.43 | newtonr | jeffspeff, See the comment by Felix in the comments section on that page. I believe that answers your question. |
| 18:02.12 | newtonr | I'm not sure if that is the best workaround.. I don't know |
| 18:02.17 | newtonr | Sounds like one option at least. |
| 18:04.18 | jeffspeff | newtonr, thanks. is the tcptls.c file something that gets compiled when making * or just something i can edit and changes apply upon saving? |
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| 18:05.46 | Obelix-1 | hello people |
| 18:06.07 | jeffspeff | hello |
| 18:06.45 | Obelix-1 | jeffspeff ... I'm try to integrate twilio in my Asterisk ... and xml it's not calling my dialler can you help me please ? ... |
| 18:06.58 | *** join/#asterisk viro (~viro@unaffiliated/viro) |
| 18:07.13 | Obelix-1 | what I need to check ... because I think I configurate everythink good ... |
| 18:07.32 | jeffspeff | i'm not sure that i can, but if you post your issue, logs and more info maybe somebody can help |
| 18:07.44 | Obelix-1 | oh okay ... |
| 18:07.51 | Obelix-1 | Thank you |
| 18:09.04 | voipmonk | twilio has to fix their documentation |
| 18:09.33 | Obelix-1 | why ? it's not working anymore true xml ? |
| 18:09.58 | voipmonk | keep at it - it works but not always as described. |
| 18:10.13 | Obelix-1 | I can see that my friend voipmonk |
| 18:10.22 | voipmonk | reading your issue again... stubby |
| 18:10.28 | voipmonk | stdby,too |
| 18:10.28 | Obelix-1 | before was working ... |
| 18:10.43 | voipmonk | they fed the mogwai last night... |
| 18:10.45 | newtonr | jeffspeff, Probably have to recompile for changes in that file |
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| 20:09.48 | BLAH_BLAH_BLAH | I was wondering if anyone had seen a way to identifying a call that possibly have been a 3way call. I've read about a 3rd party application that will monitor the call voice-payload and other variables to determine if it may be a 3way call. Is there any free applications of doing this or any scripts to run in asterisk to identify in the same manner? |
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| 20:50.58 | jkroon | does anybody know of a way to check if a certain path exists on the filesystem from dialplan, preferably avoiding SHELL() function? |
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| 21:14.17 | jeffspeff | I'm having a really hard time trying to figure out where this option is set from my configs. https://github.com/asterisk/asterisk/blob/13.10/include/asterisk/tcptls.h#L83 is that the same as tlsverifyclient and/or tlsdontverifyserver |
| 21:16.50 | jkroon | i can only find two references to that in the codebase. |
| 21:17.04 | jkroon | 1 - where you've located it, and secondly where it's tested whether the CN should be ignored. |
| 21:17.10 | jkroon | conclusion: you can't set it. |
| 21:18.24 | jeffspeff | so it's an unusable flag? |
| 21:18.29 | jkroon | looks like it yes. |
| 21:18.35 | jeffspeff | LMAO |
| 21:18.36 | jeffspeff | wow |
| 21:18.51 | jkroon | guessing you can disable verifyserver completely. |
| 21:19.32 | jkroon | but that also disables CA checking. |
| 21:19.41 | jkroon | ie, completely untrusted. |
| 21:19.59 | jkroon | just checked master too ... and definitely doesn't look like it can be set. |
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| 21:20.43 | jeffspeff | here's what i'm trying to do... i want to use secure calling. i'm following this https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial However, I don't/won't know the hostnames nor the IP's of these phones |
| 21:20.58 | jkroon | trivial to add. |
| 21:21.27 | jkroon | in main/tcptls.c - around line 1230 odd where the other tls config options are. |
| 21:21.42 | jkroon | <PROTECTED> |
| 21:21.42 | jkroon | <PROTECTED> |
| 21:21.54 | jeffspeff | so, should i be looking at the common name unusable feature or looking to use one of the other usable features like tlsverifyclient and/or tlsdontverifyserver ? |
| 21:22.01 | jkroon | example of what's there, just add a } else if ( ... "tlsignorecn") ... and set the flag there. |
| 21:22.32 | jkroon | phones? well, do you understand TLS in any way? |
| 21:22.58 | jkroon | not sure if the phones HAS to auth by X509, could just be that we auth to them, so then you can use tlsdontverifyclient. |
| 21:23.10 | jeffspeff | it encrypts traffic streams based of a key pair, one residing on the client, other on the server and the certs verify themselves agains a CA to see if they're trusted or not. |
| 21:23.16 | jkroon | which is what I would do if I had to enable TLS at this point. |
| 21:23.54 | jkroon | key pair ... well, TLS doesn't have to use a client cert, only a server cert is required, client cert is optional. |
| 21:24.04 | jkroon | not sure what the SIP implementation requires. |
| 21:24.27 | jeffspeff | ok. i'm just following instructions. lol https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
| 21:24.28 | *** join/#asterisk Micc (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 21:24.40 | Micc | I have a problem with asterisk 13.7.2 crashing every time swift runs. |
| 21:26.48 | jkroon | jeffspeff, i only know of one person that ever tried to run sip/tls. |
| 21:27.01 | jkroon | he gave up due to asterisk bugs. not sure if he revisted again. |
| 21:27.08 | jkroon | i probably should look at it at some point. |
| 21:29.15 | jkroon | and you'll need to implement srtp as well. |
| 21:39.10 | jeffspeff | yeah, i was looking into that |
| 21:39.23 | jeffspeff | i need to learn more about the difference between srtp and zrtp |
| 21:39.56 | jeffspeff | i've seen both terms tossed around pretty loosely, like they're interchangeable but i know enough to know better than that |
| 21:40.17 | voipmonk | nope |
| 21:40.20 | voipmonk | not interchangeable |
| 21:40.33 | voipmonk | does your phone support zrtp? |
| 21:42.34 | jeffspeff | not sure. i plan on using this on many different devices, so i'll go with the protocol most widely supported |
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| 23:43.37 | snadge | hmm.. so i have an issue with forwarding a call after changing the cid to private |
| 23:43.53 | snadge | asterisk seems to be setting the cid to "anonymous" .. and i haven't been able to figure it out, so im looking at the source code to chan_sip |
| 23:44.39 | snadge | the odd part is, is that works for calls from one provider.. but not another.. and i can't see anything obvious in the invite which would indicate why.. they both set Privacy: id .. odd |
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