| 00:22.53 | *** join/#asterisk knightr (~Nicolas@mythtv/developer/knightr) |
| 00:23.26 | knightr | Hi!Does Asterisk do lazy evaluation when we use the "&" operator? |
| 00:29.03 | [TK]D-Fender | As in? |
| 00:30.38 | knightr | In $["${__REVERSAL_REJECT}"="TRUE" & "${CHANNEL(reversecharge)}"="1" ] , would "${CHANNEL(reversecharge)}"="1" ] be evaluated? |
| 00:31.11 | knightr | __REVERSAL_REJECT is not "TRUE"... |
| 00:31.21 | knightr | oops, if... |
| 00:32.38 | *** join/#asterisk Dovid (~dovid@ool-45738930.dyn.optonline.net) |
| 00:39.45 | knightr | [TK]D-Fender, as in this ^ if you were talking to me... |
| 00:40.59 | [TK]D-Fender | Honestly not sure.... |
| 00:41.50 | knightr | no problem, thank you! |
| 00:42.06 | knightr | I tried find info on this but didn't find anything... |
| 00:42.30 | knightr | it looks like it *does not* do lazy evaluation but then maybe I did something wrong in my test... |
| 00:46.29 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 02:28.59 | snadge | hmm.. send call to ip function seems to be sending as UDP, but i need it to send as TCP.. stupid lync server |
| 02:29.11 | snadge | outbound is working just fine |
| 02:33.50 | snadge | do i need to use an outbound proxy for this? |
| 02:36.55 | snadge | transport=tcp seems to be an obvious option for the trunk, and the extensions.. |
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| 07:16.59 | *** join/#asterisk sparrowvivek16 (75d7b707@gateway/web/freenode/ip.117.215.183.7) |
| 07:17.11 | sparrowvivek16 | hello? |
| 07:17.17 | sparrowvivek16 | anyone here? |
| 07:17.56 | sparrowvivek16 | i need some technical support |
| 07:18.36 | WIMPy | ~ask |
| 07:18.36 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 07:19.19 | sparrowvivek16 | ok... how do i setup VPN in asterisk the openvpn service |
| 07:19.58 | WIMPy | not |
| 07:20.12 | sparrowvivek16 | i know how to do it with hamachi... but its not reliable for me... |
| 07:20.17 | WIMPy | You can setup openvpn, but that's not related to Asterisk. |
| 07:20.47 | sparrowvivek16 | how so? |
| 07:22.29 | sparrowvivek16 | i am sorry but there is openvpn service attached to asterisk 11 right? |
| 07:22.39 | sparrowvivek16 | i just need info to start it... |
| 07:22.50 | WIMPy | no |
| 07:22.58 | sparrowvivek16 | why? |
| 07:23.24 | WIMPy | They are two different applications, not related to another. |
| 07:23.26 | sparrowvivek16 | wimpy? |
| 07:24.26 | sparrowvivek16 | ok let me rephrase i want my asterisk server behind Openvpn now can you guide me: |
| 07:24.27 | sparrowvivek16 | ? |
| 07:25.22 | WIMPy | With what? |
| 07:29.02 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 07:36.43 | sparrowvivek16 | hello? |
| 07:37.01 | sparrowvivek16 | come on guys i see more that 50 people here |
| 07:38.00 | sparrowvivek16 | i want a vpn service to run myserver privately |
| 07:38.49 | sparrowvivek16 | anyone? |
| 07:38.59 | nunne | VPN has nothing to do with asterisk. You setup a private virtual network with VPN.. that Asterisk can utilize. But it has nothing to do with Asterisk. |
| 07:39.12 | nunne | For help with openvpn i should suggest you use their channel |
| 07:39.39 | nunne | if you setup net-to-net vpn it should work with asterisk without any problems. |
| 07:40.11 | sparrowvivek16 | https://www.youtube.com/watch?v=FDbs4SIkqxI&list=WL&index=27 |
| 07:40.36 | sparrowvivek16 | if you see this video there is a guy who setup asterisk with vpn |
| 07:40.47 | sparrowvivek16 | using hamachi |
| 07:41.22 | WIMPy | You don't really expect us to watch a video to find out what you might think, do you? |
| 07:44.36 | sparrowvivek16 | the ponit is he installed hamachi on both asterisk and connected to his client |
| 07:44.59 | sparrowvivek16 | i want to do it with openvpn |
| 07:46.41 | *** join/#asterisk CeBe (~CeBe@xd9beef6f.dyn.telefonica.de) |
| 07:47.09 | MrTAP | Suggest you start here: https://openvpn.net/index.php/open-source/documentation/howto.html |
| 07:48.21 | mrfrenzy | sparrowvivek16: it's like you are trying to ask Microsoft how to install a new cpu in your server. sure windows runs better with a faster cpu, but Microsoft will not help you install it |
| 07:48.41 | mrfrenzy | openvpn is a great tool, but not the topic of this channel |
| 07:49.50 | LiuYan | hmm, the 'nat=force_rport,comedia' setting works ok for normal SIP phones, but not for WebRTC client, do I need to config ICE+STUN+TURN settings to let WebRTC works in NAT environments ? |
| 07:53.34 | *** part/#asterisk sparrowvivek16 (75d7b707@gateway/web/freenode/ip.117.215.183.7) |
| 07:55.04 | nunne | LiuYan: webrtc needs ICE. so configure ICE + stun-server in rtp.conf and on the peer in sip.conf |
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| 08:04.50 | *** join/#asterisk Guest60116 (75d7b707@gateway/web/freenode/ip.117.215.183.7) |
| 08:04.59 | Guest60116 | hello |
| 08:05.01 | Guest60116 | ? |
| 08:05.24 | LiuYan | nunne: I installed & ran stun-server on the same machine where asterisk running, and add 'icesupport=yes' 'stunaddr=<the_ip>' in rtp.conf, and add 'icesupport=yes' 'transport=udp,ws,wss' 'avpf=yes' 'encryption=yes' 'dtlsenable=yes' 'dtlsverify=no' 'dtlssetup=actpass' and dtlscafile,dtlscertfile,dtlsprivatekey configs according thsi post: http://forums.asterisk.org/viewtopic.php?f=1&t=90167 |
| 08:05.42 | Guest60116 | ok why is there opnvpn service not running message in asterisk |
| 08:07.08 | LiuYan | nunne: but still, the sound seems only one way connected, i mean the callee can hear the caller, but not vice versa. and in about 10 seconds, the callee start hearing MusicOnHold |
| 08:10.10 | Guest60116 | hello |
| 08:10.11 | Guest60116 | ? |
| 08:12.25 | *** part/#asterisk Guest60116 (75d7b707@gateway/web/freenode/ip.117.215.183.7) |
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| 08:18.53 | nunne | LiuYan: the thing about MoH might be a dialplan issue? is Stun setup correctly in the client as well? |
| 08:19.08 | nunne | I don't know if it's wise to have the stun server on the same box as asterisk. |
| 08:19.31 | nunne | might think it should be hard for the server to figure out the global adress of your box when it's just connecting internally. |
| 08:19.51 | nunne | if you try, just for tryings sake, to use googles public stun servers. what are the effects then? |
| 08:23.50 | LiuYan | nunne: sorry, google is not available in china, 'thanks' to our ccp government. the MoH happened right after a warning message in asterisk console, the message: WARNING[14868][C-00000004]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10 |
| 08:26.10 | LiuYan | nunne: and, i just use wireshark captured some network traffic, and found it seems that Asterisk 13.x has an embeded stun-server, 'pjnath-2.3.0' ? |
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| 11:41.08 | knightr | Hi! Does Asterisk do lazy evaluation when we use the "&" operator? |
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| 13:03.36 | sparrowvivek16 | Asterisk Openvpn private server, client behind blocked ISP |
| 13:03.43 | sparrowvivek16 | what should i do? |
| 13:04.06 | robmal | Change sip port. |
| 13:04.25 | [TK]D-Fender | VPN. |
| 13:04.30 | [TK]D-Fender | Apparently you have that already |
| 13:04.37 | sparrowvivek16 | how do i setup |
| 13:04.54 | [TK]D-Fender | Read the instructions for whatever VPN solution you want to use |
| 13:04.58 | [TK]D-Fender | this has nothing to do with Asterisk |
| 13:05.02 | sparrowvivek16 | yeah but don't know to setup |
| 13:05.03 | [TK]D-Fender | it is basic networking |
| 13:05.28 | sparrowvivek16 | any alternatives? |
| 13:05.31 | [TK]D-Fender | Go google up a guide (there are thousands |
| 13:05.49 | [TK]D-Fender | You just said their provider blocks them |
| 13:06.03 | sparrowvivek16 | i can connect to any client on my side but UAE |
| 13:06.12 | [TK]D-Fender | So either use a technology they don't block or use something that chouls get around it like VPN |
| 13:06.26 | [TK]D-Fender | VPN is probably the only real way |
| 13:06.41 | sparrowvivek16 | ok another doubt |
| 13:07.14 | sparrowvivek16 | suppose i have a vpn server running on the same server as asterisk... |
| 13:07.34 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 13:07.42 | sparrowvivek16 | i want my client to connect directly without any client side software |
| 13:07.58 | sparrowvivek16 | how do it do that? |
| 13:08.05 | [TK]D-Fender | Everything is software |
| 13:08.20 | [TK]D-Fender | if you want one that doesn't need EXTRA software, then pick a VPN standard their OS supports. |
| 13:08.24 | [TK]D-Fender | This is all common sense... |
| 13:08.45 | sparrowvivek16 | pardon me? |
| 13:08.55 | sparrowvivek16 | VPN standard os? |
| 13:09.14 | [TK]D-Fender | VPN standard their OS supports natively |
| 13:10.14 | sparrowvivek16 | any link i can use? |
| 13:10.42 | [TK]D-Fender | google.com |
| 13:10.58 | robmal | http://www.smashbros.com/images/character/link/main.png |
| 13:11.06 | [TK]D-Fender | Go look up whatever they are using as an OS and pick one that they support and go look up the instructions on using it |
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| 13:23.27 | sparrowvivek16 | New now know how? |
| 13:23.32 | sparrowvivek16 | hello? |
| 13:25.41 | file | O.o |
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| 13:26.47 | [TK]D-Fender | What part of this are you having trouble with? |
| 13:27.11 | [TK]D-Fender | Go find out whatever standards your client's OS supports and go set up your side to match |
| 13:30.12 | sparrowvivek16 | listen... i know these are silly questions... take my POV i just need to connect with UAE without any data loss |
| 13:30.33 | sparrowvivek16 | like do i have to setup a separte VPN server? |
| 13:30.59 | [TK]D-Fender | you have to setup VPN CONNECTIVITY on your server |
| 13:31.01 | sparrowvivek16 | and need to config app using these settings? |
| 13:31.05 | [TK]D-Fender | You do not need another machine to do this |
| 13:31.18 | [TK]D-Fender | And VPN is a global networking thing, not an "app |
| 13:31.20 | [TK]D-Fender | " thing |
| 13:31.54 | sparrowvivek16 | ok say i have VPN CONNECTIVITY on my server |
| 13:32.08 | sparrowvivek16 | how my clients can reach me... |
| 13:32.32 | sparrowvivek16 | server--> VPN---> client |
| 13:32.35 | sparrowvivek16 | thats all i ask |
| 13:33.07 | [TK]D-Fender | Which is what I told you to do. |
| 13:33.25 | sparrowvivek16 | how man.. thats all i ask |
| 13:33.27 | sparrowvivek16 | please |
| 13:33.31 | [TK]D-Fender | PICK A STANDARD |
| 13:33.34 | sparrowvivek16 | i am stuck at thiis for days |
| 13:33.35 | [TK]D-Fender | SET IT UP |
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| 13:39.43 | WIMPy | can see why |
| 13:40.12 | [TK]D-Fender | Lack of proper nouns is a clear indicator |
| 13:40.38 | WIMPy | Not even understanding the questions is a bad starting point. |
| 13:41.16 | [TK]D-Fender | Evident lack of basic networking understanding and ability/willingness to do simple web-searches is another |
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| 13:41.41 | WIMPy | I see a link. |
| 13:42.29 | Darkerr | Hello everyone I have problem with periodic announce in queue. Console says that announce is played but it isn't (only moh) http://pastebin.com/BT9Vf2i3 |
| 13:42.39 | Darkerr | Can someone hit me with advice? |
| 13:43.10 | WIMPy | Does the file exist? Is it playable? |
| 13:43.30 | [TK]D-Fender | Show us the file is there and in the right condition |
| 13:43.43 | [TK]D-Fender | play it direct to prove it works |
| 13:43.44 | Darkerr | Yes i can play it with Playback() |
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| 13:44.26 | WIMPy | Crank up some debug. |
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| 16:45.18 | Ice_Strike | Can Agent and SIP Asterisk RealTime (single table) be shared to mulitple servers? |
| 16:46.27 | [TK]D-Fender | they will fight |
| 16:46.51 | [TK]D-Fender | imagine each trying to keep track of pinging it for qualify, etc. timeouts being set on each, etc. |
| 16:46.58 | [TK]D-Fender | cluster-fuck |
| 16:47.00 | Ice_Strike | Good point. |
| 16:47.26 | [TK]D-Fender | You seem dead set on this distributed model.... very wrong approach |
| 16:47.52 | Ice_Strike | I have made very different approch this time. |
| 16:48.07 | [TK]D-Fender | people using * in larger installs take SIP away from * directly and do everything via a larger registrar/proxy like SER |
| 16:48.12 | [TK]D-Fender | Kamailio, etc |
| 16:48.48 | Ice_Strike | Indeed |
| 16:56.32 | *** join/#asterisk calder4 (~aj@190.7.158.158) |
| 16:56.54 | *** join/#asterisk [373K] (~Adium@frontdoor.fiveninety.373k.net) |
| 16:58.26 | calder4 | Hi folks |
| 16:59.37 | calder4 | It is possible to use CID Lookup Source in an outbound route? |
| 17:01.40 | [TK]D-Fender | Sounds like a FreePBX question..... |
| 17:01.52 | [TK]D-Fender | Which would not be something for here... |
| 17:02.20 | [TK]D-Fender | ~freepbx |
| 17:02.20 | 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 |
| 17:02.43 | calder4 | [TK]D-Fender: Thank you. |
| 17:05.08 | *** join/#asterisk rhamnett (~rick@109.111.202.20) |
| 17:06.46 | rhamnett | Hi all, in chan_sip I can do a guest sip call like Dial(SIP/1234@somesipserver.com) I was wondering how to do the same thing in pjsip as I don't know how(or if i need) to specify an endpoint in the pjsip.conf for a basic outgoing invite |
| 17:07.08 | *** join/#asterisk BBone_ (~Thunderbi@s75-157-233-55.bc.hsia.telus.net) |
| 17:07.17 | rhamnett | I tried Dial(PJSIP/sip:1234@somesipserver.com:5060) but its not correct |
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| 17:09.05 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels |
| 17:09.24 | [TK]D-Fender | Always check the wiki for these basics |
| 17:09.29 | rhamnett | [TK]D-Fender, yep been there.. I don't understand how it fits together |
| 17:10.13 | rhamnett | I am trying to know a) do I need a trunk to do a dial out (didn't in chan_sip) and b) how to define a trunk when no auth creds needed. Thanks |
| 17:11.29 | rhamnett | [TK]D-Fender, in the last example on that page, what does the [mytrunk] entry look like? |
| 17:12.01 | [TK]D-Fender | Indeed all those samples seem to show use of a defined endpoint |
| 17:12.24 | [TK]D-Fender | With 1 showing a contact override, but not an endpoint override |
| 17:13.00 | rhamnett | Yep :) thats why I'm struggling |
| 17:17.59 | [TK]D-Fender | I'm really just not finding anything myself. So far everything points to "everything must be defined" |
| 17:18.08 | [TK]D-Fender | file: Care to chime in? |
| 17:18.17 | file | what what |
| 17:18.19 | file | reads |
| 17:18.24 | rhamnett | think i'll try and figure out how to define an endpoint |
| 17:18.26 | file | you must ALWAYS Have an endpoint |
| 17:18.28 | rhamnett | a generic one |
| 17:18.32 | rhamnett | OK that answers it then ;) |
| 17:18.37 | [TK]D-Fender | No method found to dial out via PJSIP without a defined entry of some sort |
| 17:18.44 | file | yes, a generic one is fine |
| 17:18.55 | [TK]D-Fender | just make it as such I guess... |
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| 17:21.06 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 17:21.08 | rhamnett | So I've defined this in pjsip.conf http://pastebin.ca/3156125 and i have a working [transport] definition but when I try then to use PJSIP/number@mytrunk I get a failure everyone is busy |
| 17:21.22 | file | you have to specify the endpoint in the dial line |
| 17:21.27 | file | PJSIP/<endpoint>/<SIP URI> |
| 17:21.35 | rhamnett | i if try that |
| 17:21.37 | rhamnett | i did try that |
| 17:21.54 | file | and? |
| 17:22.14 | rhamnett | same result |
| 17:22.21 | file | the result being? |
| 17:22.22 | rhamnett | think something is wrong with pjsip conf def |
| 17:22.44 | rhamnett | file, busy/congested |
| 17:22.55 | file | provide the complete console output |
| 17:23.04 | rhamnett | file, happy to privately is that ok? |
| 17:23.14 | file | I'd prefer not to |
| 17:23.27 | rhamnett | well don't want to show the sip domain so ill mask it |
| 17:23.29 | file | otherwise others can't help |
| 17:24.15 | rhamnett | 2 secs |
| 17:25.38 | rhamnett | I've got it working with a test setup and soft phone, I have respite in the mix so its probably that |
| 17:25.42 | rhamnett | thanks for the help ill figure it out |
| 17:35.21 | rhamnett | that was supposed to say Respoke not respite :) |
| 17:35.36 | rhamnett | its something to do with Respoke so thanks it seems to work without that |
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| 18:51.43 | *** join/#asterisk Demon_VoIP (~demon@ip253.net222.n37.ru) |
| 18:53.16 | Demon_VoIP | Hello. Where i can find codec choose rules for a pjsip channel? Thanks |
| 18:53.49 | Demon_VoIP | One lag: read:ilbc, write:ilbc, native:(ilbc,alaw,ulaw,gsm). Dest: read:g722, write:g722, native:(g722,speex32,speex,ilbc,alaw,ulaw,gsm). Why transcoding? |
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| 19:44.30 | bklang | Does anyone have an idea when the next Asterisk 13 certified release will come out? The wiki page says "2-4 times per year" and the last one was in April |
| 19:47.15 | [TK]D-Fender | Cert means nothing unless you're paying Digium for support |
| 19:47.27 | [TK]D-Fender | So if you're not.. don't waste you time on it |
| 20:01.44 | Demon_VoIP | What is analog of SIPDOMAIN in res_pjsip to determine domain of outbound call? |
| 20:03.19 | Demon_VoIP | analog of variable ${SIPDOMAIN} i mean |
| 20:05.07 | *** join/#asterisk tristan-mei| (gr@unaffiliated/tristan-mei) |
| 20:08.53 | Demon_VoIP | ${CHANNEL(target_uri)} i'll try :( |
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| 20:19.25 | knightr | Hi! Does Asterisk do lazy evaluation when we use the "&" operator? |
| 20:21.45 | newtonr | bklang, I'd ask malcolmd when he is available. |
| 20:23.43 | malcolmd | we don't currently know. we've been swamped to the point where we're currently putting it off. as far as bug fixes are concerned, 13.1-cert's pretty well-heeled. it doesn't have all of the features that are current in 13.5 though. |
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| 20:58.34 | Demon_VoIP | ${CUT(CUT(CHANNEL(pjsip,local_uri),@,2),>,1)} it could be easy :( |
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| 21:15.27 | malcolmd | we've got a 13.1-cert3 that's in process, but we've got a couple of things to clean up before we can push it |
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| 21:31.36 | bklang | malcolmd, newtonr: that's what I needed to know, thanks! |
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| 23:02.04 | knightr | <PROTECTED> |
| 23:02.52 | WIMPy | By now it might have been easier to look at the source. |
| 23:03.46 | WIMPy | Or you could try to ask that question in #asterisk-dev. |
| 23:13.20 | *** part/#asterisk kharwell (kharwell@nat/digium/x-ffjuanjajgtcuvpi) |
| 23:18.19 | [TK]D-Fender | It would be easier to PROVE |
| 23:18.34 | [TK]D-Fender | Do a test with a SHELL function |
| 23:18.44 | [TK]D-Fender | If it executes then you'll have your answer instantly |
| 23:18.55 | [TK]D-Fender | This shouldn't even have to be asked.... |
| 23:19.06 | [TK]D-Fender | "core show functions SHELL" <---- |
| 23:22.34 | [TK]D-Fender | heads out for the evning... |
| 23:27.05 | knightr | we can't know all as you guys do... thanks for the help... |
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