| 00:03.49 | Penguin | Inside the macro, if the user does not provide input, but instead hangs up, the call goes back to the context where the mobile phone was Dialed. The g option to the Dial keeps the call alive. Without the g option, the call dies instantly. |
| 00:04.28 | Penguin | The call needs to stay open so that the original Dial may timeout and send the incoming call to voicemail. |
| 00:05.06 | Penguin | The problem is that the call is dying in the subcontext instead of the Dial reaching timeout and going to VoiceMail(). |
| 00:05.43 | WIMPy | Does that requite sip/SipPhone to be unreachable? |
| 00:06.01 | Penguin | No, it can be reachable or unreachable. |
| 00:07.09 | Penguin | If I remove the branched Dial in the incoming context, it doesn't matter if the sip phone is alive or not, the call will go to voicemail. |
| 00:07.10 | WIMPy | Erm. |
| 00:07.27 | Penguin | It's just dialing two things. |
| 00:07.31 | WIMPy | I don't see a Return from the macro. |
| 00:07.33 | Penguin | But the second thing is a mobile. |
| 00:08.21 | Penguin | That mobile has voicemail. The mobile voicemail will answer the call to the mobile sometimes. |
| 00:08.42 | Penguin | I don't want the caller to ever hear the mobile voicemail. |
| 00:08.48 | Penguin | Hence the macro. |
| 00:09.10 | Penguin | The macro makes the mobile provide user input to take the call. |
| 00:09.28 | Penguin | Otherwise, the call should go back to asterisk's voicemail in the incoming context. |
| 00:10.05 | Penguin | If the mobile answers the call and presses 1 to accept the call, the caller is connected to the person on the mobile. |
| 00:10.09 | Penguin | No problem there. |
| 00:11.02 | Penguin | If the mobile rings too many times, exceeding the Dial() timeout value, the call goes to asterisk voicemail in the incoming context. |
| 00:11.06 | Penguin | No problem there either. |
| 00:11.42 | Penguin | But sometimes the mobile has no signal or has a dead battery, and the mobile voice mail answers, plays a short message, then hangs up the mobile. |
| 00:11.56 | Penguin | That's where the call dies instead of staying in asterisk. |
| 00:13.26 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 00:18.19 | WIMPy | Ok, I've got a dirty hack: What if you call an additional local channel that just waits int the Dail inside the macro? |
| 00:19.23 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 00:20.05 | Penguin | I suspect the same thing will happen, but I'm willing to test it anyway. |
| 00:21.13 | *** join/#asterisk mlhess- (~mlhess@drupal.org/user/102818/view) |
| 00:23.06 | *** join/#asterisk u0m3_ (~u0m3@188.25.183.231) |
| 00:24.52 | Penguin | <PROTECTED> |
| 00:24.52 | Penguin | <PROTECTED> |
| 00:24.55 | Penguin | Didn't help. |
| 00:25.11 | Penguin | Same result -- call ended after the mobile voice mail hang up. |
| 00:25.27 | WIMPy | So it kills all 3 calls? |
| 00:27.22 | Penguin | I guess it did. Call #1 is me calling the phone number which goes to incoming context. Call #2 is going to SIP phone and Local channel where there is another Dial with a macro. Call #3 is the call to the mobile. |
| 00:27.35 | Penguin | Call #3 ends when the voice mail answers then hangsup. |
| 00:27.48 | Penguin | That ends #2, which ends #1. |
| 00:27.50 | WIMPy | Ok, so all 4 call. |
| 00:27.55 | WIMPy | s |
| 00:28.14 | WIMPy | I think it's time to show us what's ahppening. |
| 00:28.28 | Penguin | Well, call #3 calls the mobile and the Wait() like you asked me to test. |
| 00:28.53 | WIMPy | Yes, so that's a total of 4 calls. |
| 00:29.02 | WIMPy | Or channels. |
| 00:29.15 | Penguin | It's at least four channels. Maybe five. |
| 00:29.24 | WIMPy | Plus the local channel. |
| 00:38.53 | Penguin | http://termbin.com/m7js |
| 00:39.13 | Penguin | The mobile voice mail answers the call... |
| 00:39.32 | Penguin | asterisk plays the sound file and then executes Read(). |
| 00:39.55 | Penguin | <PROTECTED> |
| 00:40.05 | Penguin | The voice mail is playing the short message. |
| 00:40.23 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:40.28 | Penguin | The voice mail is not able to provide input... |
| 00:40.31 | Penguin | <PROTECTED> |
| 00:40.42 | Penguin | The Read() restarts, asking for user input again. |
| 00:40.51 | WIMPy | Ok, it says the call was answered. That seems to be the issue. |
| 00:40.54 | Penguin | Then the mobile voice mail is done and hangs up the call. |
| 00:41.00 | Penguin | <PROTECTED> |
| 00:41.11 | Penguin | As soon as the mobile voice mail hangs up, it's all over. |
| 00:41.24 | WIMPy | But then it still goes to t. I don't get that bit. |
| 00:41.37 | Penguin | The g option to the Dial keeps it active. |
| 00:41.44 | Penguin | Without the g option, it dies on the spot. |
| 00:42.41 | Penguin | Now I realize that the g option allows the callee to hang up and have the dialplan progress... |
| 00:43.03 | Penguin | So I gave it other things to do after the Dial... |
| 00:43.35 | Penguin | That prevents it from returning back to the first Dial which needs to end so the call can go to asterisk VoiceMail(). |
| 00:44.39 | Penguin | I gave it a Ringing() and a Wait(60) just for fun. It rang for a full minute before it then executed exten t. |
| 00:44.45 | Penguin | So I removed that. |
| 00:44.50 | WIMPy | Ok, but I don't see why that would kill the first dial. |
| 00:45.53 | Penguin | Let me do one without the g option. |
| 00:48.19 | Penguin | <PROTECTED> |
| 00:48.19 | Penguin | <PROTECTED> |
| 00:48.38 | Penguin | And when that channel dies, the incoming channel also dies. |
| 00:49.42 | Penguin | I need to be able to keep the call alive after the mobile answers and hangs up without having provided input to the Read(). |
| 00:50.14 | WIMPy | Try to set the MACRO_RESULT right at the beginning of the macro. |
| 00:50.40 | WIMPy | I still don't see why the first dial should terminate. |
| 00:50.51 | WIMPy | Or it actually doing so. |
| 00:52.55 | Penguin | What do you suggest I set the result to? |
| 00:53.06 | Penguin | BUSY? |
| 00:53.08 | WIMPy | BUSY |
| 00:53.11 | Penguin | OK, let me test. |
| 00:53.21 | WIMPy | Just to make sure you have a default. |
| 00:53.41 | Penguin | Yeah, I'll set it default as BUSY and then if user input it 1, change it to CONTINUE. |
| 00:53.46 | Penguin | or unset it. |
| 00:53.55 | WIMPy | Yeah. |
| 00:57.00 | Penguin | <PROTECTED> |
| 00:57.00 | Penguin | <PROTECTED> |
| 00:57.10 | Penguin | Off to VoiceMail() !!!! |
| 00:57.14 | Penguin | I think that's the solution. |
| 00:57.29 | Penguin | The problem was that there was no macro result. |
| 00:58.01 | Penguin | I'm going to eat supper and then try to clean this up and test more. |
| 00:58.07 | Penguin | I think that's the solution, though. |
| 00:58.08 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 00:58.11 | Penguin | Thanks a bunch! |
| 00:58.15 | WIMPy | Great. |
| 00:58.37 | *** join/#asterisk slav3_sergal (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 01:05.14 | *** join/#asterisk arturon (~Anecchi@187.143.164.101) |
| 01:05.30 | arturon | Hello good afternoon |
| 01:05.56 | arturon | does somebody know if reverse polarity can be done in a mgcp channel? |
| 01:06.24 | arturon | I googled it with no luck |
| 01:06.38 | WIMPy | That makes no sense. |
| 01:06.50 | WIMPy | IP doesn't have polarity. |
| 01:07.07 | *** join/#asterisk rwb (~Thunderbi@65-183-151-239-dhcp.burlingtontelecom.net) |
| 01:07.15 | arturon | I'm registering Emtas devices |
| 01:07.46 | arturon | other call agents can perfom that |
| 01:08.46 | arturon | this message in a request S:XAL/rev(-) |
| 01:08.54 | arturon | with this message in a request S:XAL/rev(-) |
| 01:09.43 | Samot | Is that even supported still in the current versions of Asterisk? |
| 01:10.18 | arturon | I beleive so |
| 01:10.30 | arturon | my release of asterisk is not too old |
| 01:10.39 | Samot | What version? |
| 01:10.40 | arturon | 3 months probably |
| 01:10.49 | arturon | let me give you the rev |
| 01:11.08 | arturon | Asterisk 13.8.2 |
| 01:11.34 | arturon | it still have the mgcp channel |
| 01:12.51 | *** join/#asterisk jeffspeff (~Jeff@50.115.253.72) |
| 01:13.14 | Samot | Is chan_ccs an Asterisk module or third party? |
| 01:13.28 | arturon | I found lines like this in the config for dahdi channels can make the reverse polarity |
| 01:13.29 | arturon | answeronpolarityswitch=yes |
| 01:13.54 | arturon | but it doesn't work when I configure it in the mgcp.conf |
| 01:14.53 | Samot | You can't use settings from other channel drivers. |
| 01:14.56 | arturon | I don't know |
| 01:15.04 | arturon | do you think it could work? |
| 01:15.07 | Samot | No. |
| 01:15.18 | Samot | That's what I said "You can't..." |
| 01:15.20 | Samot | Cannot. |
| 01:15.40 | Samot | Channel driver settings are not universal. |
| 01:17.23 | arturon | yes I was just trying to see if was programed the same way |
| 01:17.37 | arturon | but I see that's not possible |
| 01:27.47 | arturon | who can tell is there is some code in mgcp module to do reversal polarity or not? |
| 01:30.41 | WIMPy | Ok, so you all like SIP, right? Can someone tell me what "SIP/2.1 200 Minimal Impact" means? |
| 01:48.41 | ChannelZ | Well, I wouldn't say I "like" SIP, I tolerate it :) |
| 01:54.51 | *** join/#asterisk tomcruise (tom@eyeswideshut.xs4all.nl) |
| 01:58.57 | *** join/#asterisk jeffspeff (~Jeff@50.115.253.72) |
| 01:59.43 | WIMPy | You must be a very tolerant person. |
| 02:02.12 | Samot | WIMPy, error codes are more important. |
| 02:02.20 | Samot | The message can be modified. |
| 02:02.38 | Samot | Generally SIP codes follow the same as HTTP codes. So a 200 is an OK. |
| 02:03.02 | WIMPy | I think it might have been the response to a BYE for a call tat was already cleared. |
| 02:03.36 | Samot | Right but don't pull your hair out of a 200 error with a unique message. |
| 02:03.41 | Samot | 200 = OK |
| 02:03.48 | Samot | 404 = No User Found |
| 02:04.07 | Samot | Etc. There are more error codes for SIP but they follow the same logic. |
| 02:04.09 | WIMPy | Yes, but it didn't look OK at all. |
| 02:04.14 | Samot | Hah. |
| 02:04.29 | Samot | OK is just the ACK. |
| 02:05.10 | WIMPy | Maybe it was just a "200 ignored - shut up". |
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| 03:46.32 | *** join/#asterisk nfsnobody (~nfsnobody@unaffiliated/nfsnobody) |
| 03:47.28 | nfsnobody | hey guys, building a new 11.25.1 box with lua support, and I'm getting the below error on startup: |
| 03:47.33 | nfsnobody | - /usr/sbin/asterisk: symbol lookup error: /usr/lib/asterisk/modules/pbx_lua.so: undefined symbol: luaL_loadbufferx |
| 03:47.50 | nfsnobody | I've gone higher verbosity, but can't get any more information... this appears to indicate to me that it maybe can't find the lua library? Any suggestions? |
| 03:50.28 | nfsnobody | I did build lua from source, maybe there's a compiler flag I needed to add to show the source directory |
| 03:52.10 | nfsnobody | I did ensure it was enabled in menuselect, but do I need to specify some sort of location in the configure? |
| 03:52.15 | nfsnobody | I can't find a list of configure options for asterisk |
| 03:52.36 | Samot | You're missing something LUA. |
| 03:52.48 | Samot | A dev package something that has the symbols. |
| 03:53.21 | nfsnobody | yeah I thought I might be, all from source, so no packages |
| 03:53.35 | nfsnobody | I've just found the --with-lua=PATH flag on configure, I think maybe it's pointing to system lua instead of the one I've compiled |
| 03:53.44 | nfsnobody | would that path be libs, or binary? |
| 03:54.26 | Samot | You set the path for lua when you installed it. |
| 03:54.39 | Samot | I don't use LUA so I have no idea where it installs. |
| 04:01.04 | nfsnobody | thanks Samot, you've put me on the right track, system lua.h and my local lua.h were in different dirs, recompiling now |
| 04:04.23 | nfsnobody | damn, same error :( |
| 04:05.43 | ChannelZ | Did you build it as a shared lib? And installed such that it put all of its headers someplace interesting as well? (this is normally the "-dev" version of a package in packaged systems) |
| 04:06.50 | nfsnobody | building from source ChannelZ, but I checked out /usr/local/includes/lua.h which it created, and it specifies the version I've compiled form source (5.3), so I ran ./configure --with-lua=/usr/local/includes |
| 04:06.58 | nfsnobody | it built, so I assume it found that header file and ran with it |
| 04:07.12 | ChannelZ | check the config.log |
| 04:07.34 | nfsnobody | strings /usr/lib/asterisk/modules/pbx_lua.so | grep liblua |
| 04:07.34 | nfsnobody | liblua-5.1.so |
| 04:07.39 | nfsnobody | that's interesting, that infers the wrong version |
| 04:07.42 | ChannelZ | and also that you ran ldconfig to update the linked library system |
| 04:08.39 | nfsnobody | ohhh, that could be my issue |
| 04:10.04 | nfsnobody | dev libraries are a bit beyond my normal skillset, I just need to run /sbin/ldconfig right? |
| 04:10.36 | nfsnobody | ldconfig -p | grep lua |
| 04:10.37 | nfsnobody | liblua-5.1.so (libc6,x86-64) => /lib64/liblua-5.1.so |
| 04:10.42 | nfsnobody | ok, definitely referencing the wrong library |
| 04:12.00 | *** join/#asterisk voipmonk (~voipmonk@138.68.2.1) |
| 04:13.27 | nfsnobody | weird, I can't find a liblua-5.3.so anywhere |
| 04:17.18 | *** join/#asterisk D30 (~asterinux@43-226-7-6.static.rise.as) |
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| 04:27.31 | nfsnobody | ok, looks like lua from source doesn't compile a .so by default, so I had to manually patch the makefile |
| 04:27.38 | nfsnobody | rebuilding asterisk now, fingers crossed |
| 04:30.03 | WIMPy | Did you ./configure --enable-shared? |
| 04:32.06 | nfsnobody | I did not |
| 04:32.09 | nfsnobody | I probably should do that :P |
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| 04:33.20 | nfsnobody | oh do you mean on lua WIMPy? it doesn't come with a configure script, just a makefile |
| 04:33.26 | *** join/#asterisk tomcruise (tom@eyeswideshut.xs4all.nl) |
| 04:33.35 | WIMPy | Huh? |
| 04:33.55 | nfsnobody | lua source doesn't come with a configure script, it just ships with a Makefile |
| 04:34.32 | WIMPy | No cmake or waf, either? |
| 04:35.02 | nfsnobody | just a Makefile and a src dir, which has another makefile and normal source files |
| 04:35.11 | nfsnobody | hmm, it still appears to be trying to use 5.1 |
| 04:35.13 | nfsnobody | strings /usr/lib/asterisk/modules/pbx_lua.so | grep "5\." |
| 04:35.13 | nfsnobody | liblua-5.1.so |
| 04:35.30 | nfsnobody | I don't get it, I'm linking it to the right header file now |
| 04:37.18 | nfsnobody | what causes it toselect liblua-5.1.so as the correct module to use? |
| 04:37.39 | nfsnobody | I have a liblua-5.3.so in the same directory, and the .h file I point the configure to specifies version 5.3 |
| 04:38.18 | WIMPy | pkgconfig? |
| 04:38.28 | WIMPy | Anyway, I'm AFK again. |
| 04:40.07 | nfsnobody | I'm reading through the asterisk configure script, it references lua 5.2 and 5.1, but not 5.3 |
| 04:40.12 | nfsnobody | configure:27253:LIBS="-llua5.1 ${pbxlibdir} -lm $LIBS" |
| 04:40.12 | nfsnobody | configure:27141:LIBS="-llua5.2 ${pbxlibdir} -lm $LIBS" |
| 04:40.21 | nfsnobody | maybe asterisk 11 doesn't have support for lua 5.3 |
| 04:57.27 | *** join/#asterisk tomcruise (tom@eyeswideshut.xs4all.nl) |
| 04:58.47 | D30 | hello everyone... i came across with webRTC.. i read some docs online but havent to tried it yet... came accross a tutorial on asterisk + webrtc using sip5ml .. now im a little bit confused as to whether asterisk is need to implement webrtc? |
| 04:58.54 | D30 | this is the tutorial i found https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 |
| 05:17.10 | *** join/#asterisk [[thufir]] (~thufir@192.157.116.163) |
| 05:25.10 | Samot | Yes, it's needed. |
| 06:02.58 | *** join/#asterisk mirela666 (~mirkob@52D9ADEB.cm-11-1c.dynamic.ziggo.nl) |
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| 06:39.01 | D30 | @Samot thanks, but can you explain why? I've seen some webRTC in youtube without asterisk sip server included and so I am considering whether to used Asterisk server or not. Can you explain benefits of using Asterisk for webRTC? |
| 06:39.19 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 06:43.29 | Samot | WebRTC is a SIP client that uses Web Sockets |
| 06:43.36 | Samot | You need a system that supports WS/WSS |
| 06:46.33 | D30 | WSS means secure websockets right? sorry not so familiar with the abbrv.. |
| 06:46.43 | Samot | Yes |
| 06:53.44 | D30 | okay thanks.. another question.. do webRTC and Asterisk supports video call? |
| 06:55.06 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 06:55.09 | Samot | Yes. |
| 06:57.50 | D30 | great thank you :) |
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| 08:17.02 | [[thufir]] | can you use webrtc with a phone? |
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| 08:46.42 | D30 | i dont know yet if we can use webrtc with a phone.. what i know is that you can call directly using your web browser so there is no need to use a soft phone.. |
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| 08:48.03 | D30 | i keep on searching about webrtc implementation and i came across with this https://appear.in/ i think they are using webrtc... |
| 08:48.44 | D30 | so the question still pops up on my head if its really needed to have an Asterisk server in place.. |
| 08:49.19 | D30 | i have a local asterisk server now and i like to play it with webrtc.. |
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| 13:17.02 | *** join/#asterisk Maver|cK (~Mav-fp@static-host119-73-118-124.link.net.pk) |
| 13:17.08 | Maver|cK | Hello |
| 13:17.18 | Maver|cK | Any Asteirsk Guru Here to Help ? |
| 13:18.02 | WIMPy | ~ask |
| 13:18.02 | 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. |
| 13:22.12 | pawiecki | Hi. For blind-transferred calls, that failed (busy, timeout), I want the caller to be connected back to the transferer (peer that transfered the call). I think of setting variable TRANSFER_CONTEXT for transferring peer, and then making custom context to handle this. Is this a good idea to do it that way? |
| 13:25.40 | Maver|cK | Iax2 DOnt Carry Callerid Dnid Information |
| 13:26.16 | Maver|cK | what variable shud i use to DialPlan for tht |
| 13:35.43 | *** join/#asterisk tomcruise (tom@eyeswideshut.xs4all.nl) |
| 13:36.00 | WIMPy | What exactely is your question? |
| 13:36.09 | WIMPy | And why are you sending empty messages? |
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| 14:56.42 | *** join/#asterisk Cuzner (~ccuzner@74.117.140.46) |
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| 15:08.09 | Maver|cK | is there any function call dont answer and play file to user press any key if press key is correct call pass to desired dialplan |
| 15:09.37 | Samot | What are you trying to do? |
| 15:10.43 | Maver|cK | i m trying to authenticate a caller |
| 15:10.53 | Maver|cK | wheather it is geniune caller or robot dialer |
| 15:11.13 | *** join/#asterisk Elwell (~elwell@freenode/staff-emeritus/elwell) |
| 15:11.19 | Samot | So you want the call to go to a recording first? |
| 15:11.26 | Maver|cK | no |
| 15:11.30 | Maver|cK | i dont want recording |
| 15:11.38 | Samot | "play file" |
| 15:11.42 | Samot | That would be a recordin. |
| 15:11.48 | Maver|cK | yeah |
| 15:11.54 | Samot | A pre-recorded file. |
| 15:12.05 | Samot | So you want the call to be answered and playback a recording.. |
| 15:12.10 | WIMPy | 'core show application Read' |
| 15:12.12 | Maver|cK | i used background function while playing a file for example press 1 to proceed a call |
| 15:12.33 | Samot | If they press the proper digit they can make it to a device? |
| 15:12.34 | WIMPy | That's another way to do it. |
| 15:12.37 | Samot | That's an IVR |
| 15:12.38 | Maver|cK | but when i press 1 it change my extenstion |
| 15:12.44 | Maver|cK | yeah |
| 15:12.50 | Samot | Yes, IVR's need a destination. |
| 15:12.55 | WIMPy | That's why Read is easier. |
| 15:12.56 | Maver|cK | i want to dial a dnid |
| 15:13.00 | Samot | So if 1 is the correct digit.. |
| 15:13.07 | Samot | You do something with it. |
| 15:13.17 | WIMPy | And you don't need an extra Application to play the file. |
| 15:13.18 | Maver|cK | yeah |
| 15:13.25 | Samot | Then when they press 1 your dialplan executes a Dial(). |
| 15:13.33 | Maver|cK | yes |
| 15:13.39 | Maver|cK | i need this one |
| 15:13.45 | Samot | It's an IVR |
| 15:13.49 | Samot | With a dial command. |
| 15:14.03 | Samot | ~book |
| 15:14.03 | 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 |
| 15:14.31 | Maver|cK | i shud see read function ? |
| 15:14.52 | WIMPy | Read is an application, not a function. |
| 15:15.03 | Maver|cK | okay |
| 15:15.09 | Maver|cK | sorry i m noob in asterisk |
| 15:15.10 | Samot | More importantly.. |
| 15:15.13 | Maver|cK | :) |
| 15:15.21 | Samot | You asked this question in #freepbx |
| 15:15.27 | Samot | Are you using FreePBX? |
| 15:16.25 | Maver|cK | no i m using asterisk |
| 15:16.35 | Samot | Straight Asterisk? |
| 15:16.37 | Samot | No GUI? |
| 15:16.37 | Maver|cK | yup |
| 15:16.40 | Maver|cK | no gui |
| 15:17.05 | Samot | OK, then asking this question in the other room wasn't productive. |
| 15:17.06 | Maver|cK | all setting are made using cli mode |
| 15:17.34 | Samot | So you send an incoming call to an IVR context. |
| 15:17.43 | Samot | That IVR context has the digit you want to match.. |
| 15:17.47 | Maver|cK | yes |
| 15:17.57 | Samot | If that digit is pressed, you use Dial() to send the call out to another DID |
| 15:18.04 | Samot | Or to a device. |
| 15:18.06 | Maver|cK | yeah |
| 15:18.08 | Maver|cK | to device |
| 15:18.25 | Maver|cK | but ivr shud no anwer |
| 15:18.26 | Samot | Well that's it. |
| 15:18.34 | Samot | What should answer then |
| 15:18.41 | Samot | What should answer the call? |
| 15:18.42 | Maver|cK | only play file |
| 15:18.45 | Samot | Something has to ANSWER |
| 15:18.53 | WIMPy | It has to answer. Otherwise it can't work. |
| 15:18.53 | Maver|cK | call will answer by the device |
| 15:18.57 | Samot | You can't play a file on an unaswered call. |
| 15:18.59 | Samot | No. |
| 15:19.02 | Samot | That's not how it works. |
| 15:19.06 | Samot | The call is answered by the PBX |
| 15:19.11 | Maver|cK | i did it using background |
| 15:19.17 | WIMPy | You can play a file without answering, but you can't read DTMF without. |
| 15:19.22 | Samot | Because an IVR answered the call. |
| 15:19.25 | Maver|cK | call come i set background file with noanswer |
| 15:19.29 | Samot | OK. |
| 15:19.38 | Samot | You have a PBX |
| 15:19.41 | Samot | That's where the calls go. |
| 15:19.45 | Maver|cK | yeah |
| 15:19.46 | Samot | The PBX decides what to do with it. |
| 15:19.52 | Maver|cK | call will go to my pstn line |
| 15:19.56 | Samot | So I call your DID, I get your PBX |
| 15:20.02 | Samot | Once I press the proper digit |
| 15:20.05 | Maver|cK | yes |
| 15:20.06 | Samot | The PBX calls your device. |
| 15:20.12 | Maver|cK | yeah |
| 15:20.16 | Samot | You answer, the PBX bridges the channels. |
| 15:20.22 | Maver|cK | yes |
| 15:20.29 | Samot | So the first thing that needs to answer is an IVR |
| 15:20.33 | Dovid | Hi. Does the Via need to stay the same in a dialog or can it change? I have an issue where a cisco sends an invite where the via is it's external IP when then send back a 100 trying, 183 with SDP then the cisco sends back a cancel but the via is the cisco's internal IP (and not external IP as before) so OpenSipS just passed it along to Asterisk instead of handling the cancel its self |
| 15:20.34 | Samot | Because you want them to press digits. |
| 15:21.07 | Maver|cK | if i answer the call billing will charge to my client |
| 15:21.26 | Samot | You can't have them press digits in a call that's not answered. |
| 15:21.26 | WIMPy | That's the way it is. |
| 15:21.54 | Maver|cK | but i tested in sip trunk i get it wht i want |
| 15:21.57 | Samot | In fact, it will cost more to the client doing this. |
| 15:22.05 | WIMPy | You can (technically) delay answering the call until the file has finished playing. |
| 15:22.07 | Samot | Because the call is being handled longer. |
| 15:22.18 | Samot | Oh he's trying to stop robo dialers. |
| 15:22.19 | WIMPy | But if your provider allows it, is another question. |
| 15:22.30 | Samot | He wants human interaction to continue the call. |
| 15:22.30 | Maver|cK | yup i want to stop robo dialers |
| 15:22.44 | Samot | Which means the call needs to be answered by something that determines that |
| 15:23.01 | Samot | Generally, the user. |
| 15:23.34 | Maver|cK | i did it in sip trunking |
| 15:23.47 | Samot | What are you talking about? |
| 15:23.47 | Maver|cK | but issue is with iax |
| 15:24.12 | Maver|cK | sip client call to my asterisk server ... i transfer call to my another asterisk server using iax2 trunk |
| 15:24.12 | Samot | Your provider supports IAX? |
| 15:24.14 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 15:24.26 | Samot | See.. |
| 15:24.31 | Samot | That's an important piece of this. |
| 15:24.41 | WIMPy | Is it? |
| 15:24.48 | Samot | Calling between two servers |
| 15:24.57 | Samot | Call being on one and the call being sent to another.. |
| 15:24.58 | Samot | Yeah. |
| 15:25.07 | Samot | For a n00b |
| 15:25.08 | WIMPy | What difference does that make? |
| 15:25.23 | Samot | It's more than just PSTN -> PBX -> Phone |
| 15:25.23 | Maver|cK | now |
| 15:25.31 | Samot | PSTN -> PBX ->PBX -> Phone |
| 15:25.31 | WIMPy | Calling a SIP device or an IAX device is not that different. |
| 15:26.08 | Samot | I didn't say IAX was the important part |
| 15:26.19 | Samot | I said TWO PBXes in the mix is the important part. |
| 15:26.28 | Maver|cK | issue is iax2 dont send callerid or dnid |
| 15:26.39 | WIMPy | Dialling another PBX usn't any different from dialling a phone, either. |
| 15:26.55 | Samot | I know. |
| 15:27.15 | Maver|cK | i just want to know what i use instead of callerid |
| 15:27.15 | Samot | But when you think the layout is PBX -> Phone, you work a certain way. |
| 15:27.36 | WIMPy | Well, there's two solutions: 1. don't mess with the extension. See above. Or 2: Save the original extension as you have to do in all cases where you mess with the extension. |
| 15:27.39 | Maver|cK | i bebug iax2 trunk is showing only called number |
| 15:27.42 | Samot | An additional PBX adds to the equation. |
| 15:27.59 | Maver|cK | debug* |
| 15:28.05 | Samot | Why do you need to use the caller ID? |
| 15:28.14 | Samot | I asked what you were trying to do... |
| 15:28.16 | Maver|cK | dnid to dial out the number |
| 15:28.29 | Samot | What number? |
| 15:28.29 | Maver|cK | i want to dialout the dnid |
| 15:28.41 | Maver|cK | the number hit to my dialplan |
| 15:28.47 | Maver|cK | 1st |
| 15:28.56 | Samot | Let's try this from the top. |
| 15:28.59 | Samot | I call your DID |
| 15:29.02 | Samot | It goes to your PBX |
| 15:29.03 | Maver|cK | okay |
| 15:29.05 | Maver|cK | yes |
| 15:29.08 | Samot | You make me press a digit |
| 15:29.11 | Maver|cK | yes |
| 15:29.14 | Samot | I press 1 and that's OK |
| 15:29.19 | Maver|cK | okay |
| 15:29.24 | Samot | Then it dials the other PBX? |
| 15:29.29 | WIMPy | closes this window for a while |
| 15:29.39 | Maver|cK | it will dial to my pstn line |
| 15:29.49 | Samot | So it's sending the call back out to the PSTN? |
| 15:29.54 | Maver|cK | yes |
| 15:30.06 | Samot | OK, so then you still use Dial() |
| 15:30.13 | Samot | And it routes back out your PSTN lines. |
| 15:30.17 | Maver|cK | but when i use background when it say press 1 .. dialed number changes to 1 |
| 15:30.23 | Samot | Right. |
| 15:30.31 | Maver|cK | i dont want tht number to change |
| 15:30.31 | Samot | But that doesn't matter. |
| 15:30.44 | Samot | Then you need to store the DID in variable. |
| 15:31.04 | Maver|cK | but call coming from iax2 trunk |
| 15:31.14 | Maver|cK | in sip trunk i can make wht i need |
| 15:31.19 | Samot | What IAX trunk? |
| 15:31.32 | Samot | Between you and your provider? |
| 15:32.24 | Maver|cK | like my client using sip he send call to my asterisk sever A .. i PAssed This Call tow my asterisk server b using iax2 trunk |
| 15:32.45 | Maver|cK | it is okay fine till now |
| 15:33.00 | Maver|cK | now my asterisk server b will asterisk please press 1 |
| 15:33.20 | Maver|cK | client press 1 but .. my extension change to 1 i want to dial the dnid number he send |
| 15:33.43 | Samot | Then you need to store it. |
| 15:33.52 | Maver|cK | how can i store tht number |
| 15:33.59 | Samot | ~book |
| 15:33.59 | 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 |
| 15:34.12 | Samot | ^ You're going to need to read up on dialplan and functions. |
| 15:34.57 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 16:04.30 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 16:12.42 | *** join/#asterisk mub (~jub@static-173-53-12-18.rcmdva.fios.verizon.net) |
| 16:19.45 | *** join/#asterisk mcflopy (5886947c@gateway/web/freenode/ip.88.134.148.124) |
| 16:19.48 | mcflopy | hello |
| 16:22.00 | mcflopy | i am new to asterisk and i use it with the freepbx frontend. i have configured some extensions and a sipgate trunk - everything works fine. now i want to include a second trunk which must be connected over another route. so sipgate trunk over 192.168.0.1 and the german telecom over 192.168.0.2 - is this possible? |
| 16:22.46 | Samot | #freepbx is the best place. |
| 16:22.49 | Samot | ~freepbx |
| 16:22.49 | 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 |
| 16:23.47 | mcflopy | oh okay - thank you :) |
| 16:23.48 | WIMPy | I'm not sure that's an Asterisk related question at all. But it's too vague to tell. |
| 16:24.19 | Samot | Well unless there is a VPN or some sort of direct connection between him and the provider, private IPs aren't going to route publicly. |
| 16:24.24 | mcflopy | WIMPy: i think this "only" needs some pf config magic... but.... i am not deep into it :) |
| 16:28.52 | igcewieling | mcflopy: If the 2nd IP is on a different subnet from the main IP of the box, then the OS routing rules will apply. If they are on the same subnet, well good luck with that. |
| 16:29.57 | WIMPy | We don't even know what these IPs belong to. |
| 16:31.23 | igcewieling | Generally the OS routing doesn't care who the IPs belong to. |
| 16:33.19 | *** join/#asterisk iheartlinux (~jwpierce3@mail.johnthecomputerguy.com) |
| 16:34.36 | WIMPy | Ok, let me rephrase: We don't even know if they are local IPs, IP of gateways or IPs of a proxy. |
| 16:35.10 | iheartlinux | blind transfer on polycom 550's: what is needed to retain/forward caller id? the originating sip/username is what is being displayed on destination ext. |
| 16:35.15 | [TK]D-Fender | WIMPy, Answered in #freepbx |
| 16:37.03 | *** join/#asterisk Oatmeal (~Suzeanne@cpe-65-185-34-151.columbus.res.rr.com) |
| 16:40.04 | *** join/#asterisk Cuzner (~ccuzner@74.117.140.46) |
| 16:48.52 | mcflopy | so i solved it with a simple route command :) |
| 16:48.58 | mcflopy | route add -net 217.0.0.0/13 gw 192.168.0.5 dev eth0 |
| 16:50.12 | [TK]D-Fender | iheartlinux, Then you aren't doing a blind transfer |
| 16:51.19 | iheartlinux | [TK]D-Fender: then what am I doing? |
| 16:52.05 | [TK]D-Fender | IffffSo far not confirming the exact steps and methodology |
| 16:52.48 | [TK]D-Fender | because a proper one jsut works |
| 16:53.21 | WIMPy | Usually even a blone transfer just works. |
| 16:53.26 | WIMPy | +d |
| 17:04.49 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 17:04.49 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 17:09.04 | iheartlinux | well for some reason the receiving ext is showing the sip id |
| 17:20.45 | *** join/#asterisk miralin (~Thunderbi@194.8.128.47) |
| 17:23.18 | igcewieling | like this? 74afe02a609ec5e3512e35ec6a89d009@208.22.22.150:5060 |
| 17:23.39 | igcewieling | ah, sup username, not sip ID |
| 17:24.16 | *** join/#asterisk chris1980 (~chris@host86-177-212-20.range86-177.btcentralplus.com) |
| 17:26.23 | igcewieling | Polycom phones can be told how/where to get the incoming callerid info. |
| 17:26.54 | iheartlinux | some freeking module I wasn't loading. went back to autoload. |
| 17:29.15 | igcewieling | voIpProt.SIP.CID.sourcePreference: Specify the priority order for the sources of caller ID information. The headers can be in any order. If Null, caller ID information comes from P-Asserted-Identity, Remote-Party-ID, and From in that order. The values From,P-Asserted-Identity, Remote-Party-ID and P-Asserted-Identity,From, Remote-Party-ID are also valid. |
| 17:40.49 | *** join/#asterisk [[thufir]] (~thufir@192.157.116.163) |
| 17:40.52 | *** join/#asterisk jkroon (~jkroon@uls-154-73-32-15.wall.uls.co.za) |
| 17:41.16 | chris1980 | hi, ive been following the tutorials and managed to get a basic call working between two softphones on mobile handsets working, however ive tried to introduce secure calling but I get unknown ca and failiure to register 503 error, im stumped as how to resolve this as others seem to had had success with the same |
| 17:52.20 | *** join/#asterisk tomcruise (tom@eyeswideshut.xs4all.nl) |
| 18:04.20 | igcewieling | Try: https://www.google.com/webhp?q=asterisk+tls+503 |
| 18:04.56 | *** join/#asterisk BlackMaria (~BlackMari@dsl-104.163.140-101.ebox.ca) |
| 18:09.44 | Samot | If it's a self-signed cert, you need to put the cert and ca on the device. |
| 18:17.35 | chris1980 | ive tried multiple methods, resetting the keys and retransferring them to the devices, tried just using the same host/server, im going to try using openssl instead of the ast script |
| 18:18.36 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 18:19.20 | *** join/#asterisk tomcruise (tom@eyeswideshut.xs4all.nl) |
| 18:23.00 | Samot | 503 is a service unavailable error. |
| 18:23.25 | Samot | Is this Chan_SIP or Chan_PJSIP? |
| 18:26.36 | chris1980 | chan_sip |
| 18:28.34 | chris1980 | the asterisk console gives error tlsv1 alert unknown ca and warning file * open failed |
| 18:29.30 | chris1980 | but when sip reload is run it finds both the key and the crt with no warnings |
| 18:30.01 | Samot | Do: asterisk -rx "sip show settings" |
| 18:30.03 | Samot | ~pb |
| 18:30.03 | infobot | i guess pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 18:30.10 | Samot | ^ give the pb link. |
| 18:35.35 | chris1980 | http://pastebin.com/h8SSdrF1 <- link from the search bar after clicking create paste |
| 18:37.53 | igcewieling | The worst thing about Asterisk's TLS support is the total lack of useful error messages for some common problems. |
| 18:38.15 | igcewieling | for example, the actual file which failed to open. |
| 18:40.21 | *** join/#asterisk miralin (~Thunderbi@194.8.128.47) |
| 18:40.58 | chris1980 | it took me a while to figure out where to put the actual tls config aswell, the tutorial wasnt to clear about where in sip.conf to actually put it |
| 18:51.00 | *** join/#asterisk thunfisch (~thunfisch@unaffiliated/jkoppe) |
| 18:59.32 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 19:01.09 | Samot | Allowed transports: UDP |
| 19:01.45 | Samot | Nm...that's not for this. |
| 19:03.46 | Samot | Show your sip.conf and the TLS settings. |
| 19:08.43 | chris1980 | http://wwww.pastebin.com/Apzs00VN |
| 19:17.19 | Samot | And those are owned by the user that Asterisk runs as? |
| 19:17.25 | Samot | So it can open and read those files? |
| 19:20.56 | igcewieling | can you use s_client (or openssl client) to connect to tcp/5061 check the certificates and keys? |
| 19:22.21 | chris1980 | im running asterisk with sudo, which i had to use for the ast script also, im pretty new to linux so im not sure if the permissions are correct |
| 19:23.05 | Samot | ls -l /etc/asterisk/keys |
| 19:23.07 | Samot | ~pb |
| 19:23.07 | infobot | i heard pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:24.18 | igcewieling | https://www.feistyduck.com/library/openssl-cookbook/online/ch-testing-with-openssl.html |
| 19:25.16 | chris1980 | http://www.pastebin.com/g77egsZ4 |
| 19:27.45 | Samot | What user does Asterisk run as? |
| 19:28.22 | chris1980 | i run it from the chris but I have to use sudo otherwise it does not work |
| 19:28.32 | Samot | sudo to what? |
| 19:28.37 | Samot | To root? |
| 19:28.39 | igcewieling | you sudo to 'chris' |
| 19:28.45 | chris1980 | sudo asterisk -rvvvvv |
| 19:29.01 | Samot | That's the CLI/console |
| 19:29.10 | igcewieling | so you are *not* running it as user chris. |
| 19:29.17 | chris1980 | i run service asterisk start to run it |
| 19:29.18 | Samot | So basically it's running as root. |
| 19:29.23 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 19:29.33 | Samot | You do "sudo service asterisk start"? |
| 19:29.42 | Samot | Or just "service asterisk start"? |
| 19:30.12 | igcewieling | you tell what user for asterisk to run as in asterisk.conf or -u and -g. don't use sudo |
| 19:30.14 | chris1980 | when i run service asterisk start it asks for my passwrd |
| 19:30.51 | chris1980 | but to get to the cli i have to use sudo otherwise it wont run |
| 19:31.08 | igcewieling | chris1980: getting to the CLI and starting Asterisk are two totally different things. |
| 19:31.22 | chris1980 | i know, I just wanted to let you know |
| 19:31.41 | igcewieling | sudo to root. |
| 19:32.05 | Samot | Well outside of ownership permissions... |
| 19:33.12 | Samot | rw only by the owner isn't going to work. |
| 19:34.26 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 19:35.27 | chris1980 | so i should make the asterisk keys owned by root perhaps ? apologies |
| 19:35.50 | Samot | Whatever user that Asterisk is truly running as. |
| 19:36.04 | igcewieling | chris1980: If you start Asterisk as root, then you won't have this issue. |
| 19:36.11 | Samot | There's that. |
| 19:36.48 | igcewieling | then once Asterisk does all the "rooty" stuff, then Asterisk will change its userid to whatever is configured in asterisk.conf and/or the -u and -g options. |
| 19:36.57 | Samot | But the permissions should be at least 644, probably 664. |
| 19:37.22 | igcewieling | Samot: Nothing sane will start with a world readable key. |
| 19:38.01 | Samot | OK. So at least 640 |
| 19:38.05 | lvlinux | Hi everyone! Can i stop asterisk from regenerating DTMF when it detects a feature code? In other words if a feature uses something like the star key, can I prevent asterisk from sending that star to the other end of the call and just process the feature? |
| 19:38.23 | igcewieling | in order to run asterisk -rvvv the file /var/run/asterisk/asterisk.ctl needs to be read/writable by the user running asterisk -rvvv |
| 19:38.47 | igcewieling | lvlinux: as far as I know, that is the default. |
| 19:39.12 | lvlinux | igcewieling: hmmm, well it doesn't seem to be that way... |
| 19:39.29 | igcewieling | lvlinux: maybe asterisk isn't detecting the feature code? |
| 19:39.51 | lvlinux | I have a feature code set for * and the other end is catching the * and doing it's own thing. (the feature plays back a file, which asterisk does as it should) |
| 19:40.07 | *** join/#asterisk WIMPy (~wimpy@85.183.95.26) |
| 19:40.08 | igcewieling | lvlinux: does * in fact playback a file? |
| 19:40.16 | lvlinux | yes |
| 19:40.55 | igcewieling | Weird. Maybe nobody tried using * as a code instead of *X or *XX before. |
| 19:41.09 | igcewieling | does the same happen with # or even "5"? |
| 19:41.13 | lvlinux | (it's a voicemail system on the other end, with dtmf codes for functions, so I need it to not hear any dtmf from my side when the feature code is pressed) |
| 19:41.31 | lvlinux | i didn't use a # or single digit, but I did try * and then a digit. |
| 19:41.45 | lvlinux | same result (asterisk played the file as it should but the other end reacted to the code too |
| 19:41.53 | lvlinux | let me try just with # or a single digit |
| 19:42.37 | igcewieling | in any case, if Asterisk sees the digits as a feature code, then it should handle it locally and not pass the digits on. Otherwise nothing would work right in asterisk. |
| 19:42.53 | lvlinux | that's what I thought shoud happen. |
| 19:43.07 | igcewieling | paste your features.conf, maybe someone will want to dig into it. |
| 19:44.40 | lvlinux | single digit does same thing. |
| 19:44.55 | Samot | Show the call. |
| 19:45.00 | lvlinux | let me check over my features.conf first and make sure I didn't mess with something years ago... |
| 19:45.04 | Samot | Show what the dialplan is doing with the call.. |
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| 19:55.36 | lvlinux | here you go: http://pastebin.com/wF3yWZ3R |
| 19:57.24 | igcewieling | I don't see a sequence to match a single * |
| 19:57.50 | lvlinux | i changed it to 9 |
| 19:57.53 | igcewieling | It is very possible Asterisk requires a min of 2-digit feature codes. |
| 19:58.20 | lvlinux | but it worked fine---Asterisk played the file as it should with both double and single digits |
| 19:59.08 | igcewieling | if it passed the DTMF, then it is NOT working fine. |
| 19:59.28 | igcewieling | you could try asking on the asterisk-users mailing list or maybe file a bug report. |
| 19:59.32 | lvlinux | well i mean that it did recognize a single digit at a feature code and processed it. |
| 19:59.45 | lvlinux | s/at/as/g |
| 20:03.18 | Samot | Wait, this is because when you're on the call the other party hears you pressing DTMF? |
| 20:07.02 | lvlinux | yes |
| 20:10.20 | lvlinux | i want the other side to hear nothing but the file being played back |
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| 20:21.00 | Samot | Hrm.. |
| 20:21.15 | Samot | I think the issue is playbackfile => 9,caller,Macro,playbackfile |
| 20:21.24 | Samot | I think caller should be "self" or "peer" |
| 20:22.49 | lvlinux | k i'll try that |
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| 20:27.59 | Samot | Who's supposed to hear the playback? |
| 20:28.34 | lvlinux | both sides preferrably |
| 20:28.47 | lvlinux | peer seemed to work, only i couldn't hear it, but that's ok |
| 20:29.22 | lvlinux | thanks for the tip! |
| 20:29.36 | Samot | You can try self/peer |
| 20:29.45 | lvlinux | you mean both with the slash? |
| 20:30.00 | Samot | Yes. |
| 20:30.05 | lvlinux | oh yes i see it in the help |
| 20:30.18 | Samot | self = channel that initiates the feature |
| 20:30.22 | lvlinux | i forgot about that---i was thinking there was only activated by |
| 20:30.22 | Samot | peer = other channel |
| 20:30.39 | Samot | ActivatedBy is no longer honored. |
| 20:31.17 | lvlinux | huh? Then wouldn't it ignore the "peer" in "self/peer"?? |
| 20:31.27 | lvlinux | ;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,<AppArguments>[,MOH_ |
| 20:31.45 | Samot | ActivatedBy is no longer honored. The feature is activated by which channel DYNAMIC_FEATURES includes the feature is on. |
| 20:31.48 | igcewieling | the most you should expect to hear on the other end is a short chirp when Asterisk intercepts DTMF. |
| 20:32.01 | Samot | Use predial to set different values of DYNAMIC_FEATURES on the channels. Historic values are: "caller", "callee", and "both". |
| 20:32.53 | lvlinux | yeah that's right (it's been a while since I messed with features) |
| 20:33.16 | Samot | I think you need self/peer because there's no "both" option. |
| 20:33.34 | igcewieling | Looks like someone didn't read features.conf.sample |
| 20:33.49 | lvlinux | Samot: so it would use both as options to ActivateOn? |
| 20:33.56 | Samot | I think so. |
| 20:34.14 | Samot | Haven't done it, I'm just basing this on what I see in the Wiki. |
| 20:34.17 | Samot | And what I would do. |
| 20:34.22 | Samot | Which is set it and test it. |
| 20:34.56 | lvlinux | igcewieling: I read it, that's why I'm confused! lol |
| 20:35.19 | lvlinux | The line I pasted above is from the sample file |
| 20:35.36 | lvlinux | and still lists the ActivatedBy. |
| 20:35.42 | Samot | So does the Wiki |
| 20:35.51 | Samot | But the Wiki also has the lines I pasted. |
| 20:37.42 | igcewieling | which wiki? https://wiki.asterisk.org/wiki/display/AST/Home or voipinfo,.org 8-) |
| 20:38.38 | igcewieling | the first wiki should be the one of the first places you look. the second one should be the last place you check. voip-info.org is outdated and disorganized. |
| 20:39.28 | Samot | The Asterisk Wiki is always the first place I look |
| 20:41.51 | [TK]D-Fender | the sample configs should be the first place.... |
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| 21:17.38 | Samot | lvlinux: Did self/peer work? |
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| 23:55.36 | funxion | does anyone why a call would not hunt through the codecs between asterisk and freeswitch? |