| 00:00.24 | vitorbaptista | These are my iax.confs: http://pastebin.com/pBy5rjNa |
| 00:00.55 | vitorbaptista | robmal: I've been using type=friend on both sides. It used to work, but I've done some change and it stopped working (not sure what) |
| 00:11.16 | vitorbaptista | Duh! Nevermind: the issue was with the username. Read it many times but didn't notice an extra dash at the end. |
| 00:11.18 | vitorbaptista | o/ |
| 00:11.41 | robmal | :-) |
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| 05:25.40 | overyander | if i have sip.conf set to only use tcp, my RTP will still be transmitted in UDP right? or does RTP switch to TCP as well? |
| 05:26.06 | WIMPy | RTP stays UDP |
| 05:26.18 | overyander | ty |
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| 05:27.06 | [TK]D-Fender | rtp is always UDP |
| 05:29.23 | WIMPy | I think RTP over TCP does exist, but Asterisk doesn't support it and it is most probably a really bad idea anyway. |
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| 09:11.29 | marksm2 | Hi, I want to use asterisk 11 withConfBridge for conference calls. is it possible to do the following: create an extension ie 100, when an user dials it, it dials a few extensions eg 21,22,23,24 and when anyone of them picks up the call they are automatically joined in the conference call? |
| 09:12.20 | robmal | Use originate. |
| 09:22.05 | marksm2 | robmal: thanks. do you know if there is some example online which creates a simple confbridge and uses originate to map an extension to it so that once called, it dials out to a fixed list of SIP accounts |
| 09:25.37 | robmal | Yes, but it's mine and you'll NDA and other crap to see the code :-) |
| 09:25.49 | robmal | Nothing that hard, seriously. |
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| 09:29.38 | marksm2 | robmal :) so basically do I need to setup a ConfBridge menu which performs addcaller like here? https://wiki.asterisk.org/wiki/display/AST/ConfBridge |
| 09:29.57 | marksm2 | robmal: or can I skip setting up a menu? |
| 09:30.40 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 09:39.24 | robmal | You can skip it, just put it in the dialplan. |
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| 10:22.50 | marksm2 | robmal: thanks. do you know if I can put multiple SIP/number&SIP/number2 in Originate() ? |
| 10:23.56 | marksm2 | robink, if I put 2 SIP in Originate() it does not work, if I put only one SIP/number then it calls that number and put him in the conference but it hangs up on me as the caller and tells to him "you are the only partecipant in the conference" |
| 10:24.28 | marksm2 | robmal, robink was meant for you. autocomplete :) |
| 10:24.52 | robmal | SIP/123&SIP/234 |
| 10:25.08 | robmal | Or just use multiple originates. |
| 10:29.51 | marksm2 | robmail: thanks, this is my whole small extensions.conf. I call from SIP/21 extension 100. if I put only SIP/22 in Originate() then it calls 22 and puts him in the conference but hangs me up. if I put SIP/22&SIP/21 it does nothing.probably an error. see my dialplan here: http://pastebin.ca/3166072 |
| 10:31.18 | robmal | Put ext100 in conferences. |
| 10:42.50 | marksm2 | robmal, thanks. it still does not work, see updated asterisk dialplan and debug output. thanks a lot for your help! http://pastebin.ca/3166084 |
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| 10:52.32 | robmal | My bad, originate can't into multiple destinations. |
| 10:52.36 | robmal | Dial can. |
| 10:53.43 | marksm2 | robmal, so do I need to use multiple Originate() , same paremeters just different SIP/number, therefore a sequence of Originate() ? How can I avoid that I as the caller to 100 get joined to the conference too? |
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| 10:55.17 | robmal | Put the conference on hold and call ext100? |
| 10:56.01 | robmal | I've done it by AMI, with scheduling ahead and calling the parties. I think it's the better way. |
| 10:58.09 | marksm2 | robmal: thanks! I need to call a few people dynamically and each one that picks up gets put in the conference including me. this saves us time. we cannot plan calls ahead. So how could I put the conference on hold and call ext100 in my case? do you happen to know what commands I need to use? I'm a bit rusty with asterisk. |
| 10:59.35 | robmal | This is a bit trickier, i've made a feature code which creates a dynamic confroom, puts you in it and uses channelredirect for the other party. |
| 11:01.19 | marksm2 | robmal, I need only 1 confroom so I can use static confbridge numbers. do you happen to know where I need to look for docs (which asterisk commands) to do the stuff you said? thanks! |
| 11:02.08 | robmal | Asterisk wiki has all the answers, ignore the voip-info links, that's old and sometimes harmful. |
| 11:02.49 | marksm2 | robmal, thanks I will dig into it. but any starting pointers ? in the sense of keywords to look for. the wiki is big |
| 11:03.07 | robmal | One sec. |
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| 11:10.10 | robmal | Ok, so i do it this way: |
| 11:10.18 | robmal | make a confroom (confbridge/meetme) |
| 11:10.33 | robmal | channelredirect channel |
| 11:10.40 | robmal | channelredirect bridgepeer |
| 11:11.29 | robmal | There are some quirks, like if you don't check if you already have a call on hold to that confroom you'll have two connections, but i think you'll figure it out. |
| 11:13.23 | marksm2 | robmal, thanks. so basically I will lookup these keywords on the wiki, right? one more question: when I use 2 Originate() in sequence to eg. call SIP/22 and then SIP/23, it calls first 22 then only after 22 picked up, it calls 23. is it possible to call everyone in parallel to speed up things? Ive read there is an Async option but it was in the Asterisk Manager API Action Originate |
| 11:14.05 | robmal | Use dial, but that won't put them in the confroom by itself. |
| 11:15.05 | marksm2 | robmal, ah you mean first use Dial() and then use other cmds to redirect them to the conference? with channel redirect? |
| 11:19.11 | robmal | Yes. |
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| 11:25.48 | marksm2 | robmal: thanks. I see ChannelRedirect(channel,[context,[extension,]]priority) in that case if first do exten => 100,1,Dial(SIP/22&SIP/23,,Tt) |
| 11:25.49 | marksm2 | <PROTECTED> |
| 11:27.48 | robmal | Well, the channel you want to redirect, so first your channel from $CHANNEL, than $BRIDGEPEER. Depends who should be the first in the confroom for about a second. |
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| 11:54.22 | xpheres | hello, I'm trying to establish a connection using SRTP encryption, but I'm stucked in the process |
| 11:54.28 | xpheres | anyone has succeed doing it? |
| 11:54.50 | robmal | Yes. |
| 11:55.13 | xpheres | I'm connecting with zoiper |
| 11:55.24 | xpheres | whatever I do, I'm not able to log in or to make a call |
| 11:55.42 | robmal | core set verbose 10 |
| 11:55.47 | robmal | core set debug 10 |
| 11:55.56 | xpheres | ok |
| 11:56.04 | robmal | Copy and paste it somewhere. |
| 11:57.09 | xpheres | an amateur question |
| 11:57.20 | xpheres | should I open a different port on the router when using tlp instead of udp? |
| 11:58.12 | robmal | Well, that's the whole point, so, yes. |
| 11:58.25 | xpheres | mm |
| 11:58.36 | xpheres | can you help me with that? |
| 11:58.46 | xpheres | I only have options : TCP, UDP ESP , GRE |
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| 12:02.50 | xpheres | http://pastebin.com/W0wjzUd1 |
| 12:03.07 | marksm2 | robmal. thanks for your help. I tried to issue a exten => 100,1,Dial(SIP/22,10,Tt) and then exten => 100,2,Verbose(1,${CHANNEL}); on the console I see the Verbose cmd output only when 22 does not pick up within 10secs, otherwise Verbose is not executed. I thought the exten cmds usually gets executed in sequence, from ringing, pickup to call hangup. any idea if I did something wrong? |
| 12:05.18 | xpheres | to stop verbosity core set verbose 0? |
| 12:05.35 | xpheres | ok yes |
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| 12:23.10 | marksm2 | robmal: do I have to use the G option in Dial() to execute the next cmds in the externsion in order to call ChannelRedirect() ? |
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| 12:37.28 | qakhan | there are 5 phones in an office. joe phone ext is 8000 and receives call on. he also works in other department and his other ext is 8005, he needs to receive calls coming on ext 8000 on his ext 8005. |
| 12:39.23 | qakhan | basically he can be at any ext from 8000 to 8010. he just dial some number on current ext so he start receiving call of 8000 ext |
| 12:55.17 | robmal | Sorry, i had to fix my 'insider program' issues with w10 == no wifi |
| 12:57.01 | robmal | marksm2: I didn't get it right the first time either ;-) If you want my ready to use solution you'll need paypal ;-) |
| 12:57.40 | robmal | And seriously, it's not that hard, just keep poking around. |
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| 13:36.22 | [TK]D-Fender | marksm2, Dial Exits when it is done. Either by the call completing or never getting established. "G" has no impact on AMI redirection. |
| 13:41.42 | marksm2 | [TK]D-Fender I am now using Dial( .. G(mycontext^myext^otherprio) then at pos otherprio I use a GoTo() for the caller and at otherprio+1 a Goto() for the called party. and then I do a channel redirect to the ConfBridge extension. it works with 2 parties but not with 3 yet |
| 13:42.20 | [TK]D-Fender | I think I got the wrong G there.... |
| 13:42.32 | [TK]D-Fender | Ok, so you are doing a forced Goto on answer? |
| 13:42.37 | [TK]D-Fender | that is still just 2 parties. |
| 13:42.45 | [TK]D-Fender | Where do you get 3? |
| 13:43.45 | marksm2 | [TK]D-Fender I would like to create an extension with ConfBridge which you can call and then it calls a predefined number of SIP extensions and joins everyone in the conference room automatically |
| 13:43.51 | [TK]D-Fender | Scroling back it looks like you just want to Originate to a pile of people.... |
| 13:44.15 | [TK]D-Fender | At this point you shouldn't be using G() at all and just dial them independently and toss them into the conference |
| 13:44.27 | [TK]D-Fender | You cn do them all at the same time |
| 13:45.26 | marksm2 | [TK]D-Fender: whats cn? I just Dial(SIP/num1&SIP/num2) but without G() dial does not exit until hangup, so I dont know how to toss them to the conference. any ideas? |
| 13:45.57 | [TK]D-Fender | can |
| 13:46.09 | [TK]D-Fender | Don't mutli-dial. |
| 13:46.21 | [TK]D-Fender | do MULTIPLE originates each to its own target |
| 13:46.41 | marksm2 | [ |
| 13:46.52 | [TK]D-Fender | because when you multi-dial only one is going to answer and as soon as it does the others get dropped |
| 13:47.36 | marksm2 | [TK]D-Fender: I tried with multiple Originate() but then it first call SIP/num1 and blocks until SIP/1 picks up. only then it executes the 2nd Originate(). any idea how to do it asynchronously? thanks |
| 13:48.14 | [TK]D-Fender | show us |
| 13:48.28 | [TK]D-Fender | what do you mean Originate()? You're not talking about the DIALPLAN APP are you? |
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| 13:51.45 | marksm2 | [TK]D-Fender: I did like this http://pastebin.ca/3166084 but I used 2 Originate() in sequence , the first with SIP/22 the 2nd with SIP/21 (since Originate() does not allow multiple SIP/number) but then with 2 Originate() it dials the first sip num. and only after the first sip ext picked up it dials the 2nd one. I need dialing everyone in parallel |
| 13:52.14 | [TK]D-Fender | Wrong solution |
| 13:52.21 | [TK]D-Fender | Do NOT use the dialplan app for this, it IS blocking |
| 13:52.31 | [TK]D-Fender | use AMI Originate or call-files |
| 13:53.17 | marksm2 | [TK]D-Fender, thanks a lot. but can I use AMI from the dialplan? or do I need to call an external app? |
| 13:53.32 | [TK]D-Fender | clearly external |
| 13:53.38 | [TK]D-Fender | Or shell out to call the CLI Originate |
| 13:53.49 | [TK]D-Fender | (also gotten to externally) |
| 13:54.12 | [TK]D-Fender | exten => 100,1,Originate(SIP/22&SIP/21,exten,conferences,200,1) <- also if you read the instructions you can't put multiple on this. That is not a mistake you should have left in there. |
| 13:54.58 | marksm2 | [TK]D-Fender, yes I know now that it takes only one SIP/number argument. I just pasted it because I already had the link on pastebin without creating a new one. |
| 13:55.17 | [TK]D-Fender | System(asterisk -rx "channel originate ......" &) |
| 13:55.21 | [TK]D-Fender | ^ |
| 13:56.02 | marksm2 | [TK]D-Fender, thanks. so you think System() is the easiest way to go in my case? keep in mind that I would like the caller be put in that conference too. not only the called parties |
| 13:56.16 | marksm2 | or are call files more flexible? |
| 13:56.47 | [TK]D-Fender | they are separate calls. |
| 13:57.00 | [TK]D-Fender | they have NO relation to whatever told them to originate |
| 13:57.11 | [TK]D-Fender | So launch all the others and then CONTINUE into the conference |
| 13:58.12 | marksm2 | [TK]D-Fender, for example with Dial() and ChannelRedirect() I am able to call (only 1 party), put him into the conference and myself too. but since you said Dial() with multiple arguments hangs up on others. that approache does not work |
| 13:58.48 | [TK]D-Fender | And I told you to originate them all independently NOT using the dialplan app ver of that function |
| 13:59.05 | [TK]D-Fender | So launch them all off and go into the conf |
| 14:00.36 | marksm2 | [TK]D-Fender thanks alot. but as the caller (myself) to go into the conf can use ChannelRedirect() with your method too? (first System(..) then ChannelRedirect() |
| 14:00.59 | [TK]D-Fender | .... |
| 14:01.05 | [TK]D-Fender | You don't NEED to redirect. |
| 14:01.15 | [TK]D-Fender | You the call just issed all of those originates. |
| 14:01.24 | [TK]D-Fender | JUST CALL CONBRIDGE RIGHT AFTER THEM ALL |
| 14:01.33 | [TK]D-Fender | It's the next dumb app to call. |
| 14:01.58 | marksm2 | [TK]D-Fender, I understand now, thanks! :) I'll try it right now |
| 14:02.08 | [TK]D-Fender | System(asterisk -rx "channel originate ......" &) |
| 14:02.13 | [TK]D-Fender | Confbridge() |
| 14:02.15 | [TK]D-Fender | The end |
| 14:09.44 | qakhan | there are 10 phones in an office. joe phone ext is 8000 and receives call on. he also works in other department and his other ext is 8005, he needs to receive calls coming on ext 8000 on his ext 8005. |
| 14:09.44 | qakhan | basically he can be at any ext from 8000 to 8010. he just dial some number on current ext so he start receiving call of 8000 ext |
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| 14:25.44 | marksm2 | [TK]D-Fender, you are genius! your solution works perfectly. thanks to robmal too. |
| 14:27.39 | [TK]D-Fender | You're welcome |
| 14:28.09 | [TK]D-Fender | qakhan, If you set your dialplan up to do that |
| 14:29.10 | qakhan | [TK]D-Fender how? |
| 14:29.36 | [TK]D-Fender | qakhan, `It's YOUR dialplan. make something they dial that saves that decision somewhere. In your OTHER parts of the dialplan CHECK for it |
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| 14:30.42 | qakhan | [TK]D-Fender got it |
| 14:32.10 | qakhan | get 8000 ext from desire phone and save that phone callerid it in veriable. if call comes for 8000 send it to saved callerid |
| 14:32.12 | qakhan | right? |
| 14:32.30 | [TK]D-Fender | to whatever |
| 14:32.35 | [TK]D-Fender | save it however you want |
| 14:32.46 | [TK]D-Fender | determine the caller whichever way you want |
| 14:33.37 | qakhan | ok thanks :) |
| 14:37.50 | marksm2 | [TK]D-Fender, one thing about those conference rooms is if lets say a group of 5 people gets called. all pickup and 2 of them forget to hangup. then they put the phone in their pocket and the conf call continues but they result busy if someone needs to call them. I'm thinking some way to end the conference automatically. perhaps with a time limit? is that possible with asterisk in the case of conf calls? |
| 14:38.47 | [TK]D-Fender | Put a limit on the call itself beofre the conf. |
| 14:38.58 | [TK]D-Fender | "core show function TIMEOUT" <- |
| 14:49.52 | marksm2 | [TK]D-Fender, thanks for the timeout var hint. it works well! |
| 14:50.09 | [TK]D-Fender | np |
| 14:50.14 | [TK]D-Fender | heads out for a few hours |
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| 16:28.30 | qakhan | [TK]D-Fender this is how i did http://pastebin.com/m72R1akL |
| 16:28.47 | qakhan | do you think it is right way to do |
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| 17:57.12 | xpheres | hello,I've been trying to configure extensions to allow SRTP encryption without success, I wonder if someone can guide me |
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| 18:03.09 | DarkHelmet | cool, got mdc1200 working to STUN radios |
| 18:15.05 | [TK]D-Fender | qakhan, So you ONLY want 8000 to do this? And any extension to be able to dial * to set the redirect? |
| 18:16.00 | xpheres | anyone knows if cisco voip telephones 7940 7960 or 7970 support SRTP encryption? |
| 18:23.41 | qakhan | [TK]D-Fender yes |
| 18:24.19 | [TK]D-Fender | qakhan, It'll work... except it won't survive an * restart |
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| 18:24.34 | [TK]D-Fender | qakhan, Globals die with * and have no starting value |
| 18:24.45 | qakhan | yes i know. |
| 18:24.46 | [TK]D-Fender | qakhan, AstDB is what you should be doing this with |
| 18:25.27 | qakhan | [TK]D-Fender they are good with that. |
| 18:25.50 | [TK]D-Fender | qakhan, not a good choice sine AstDB isn't even 1 line extra |
| 18:25.53 | [TK]D-Fender | since* |
| 18:26.20 | qakhan | by defualt 8000 will receive the call |
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| 18:26.56 | qakhan | you are right, |
| 18:27.29 | [TK]D-Fender | <qakhan> by defualt 8000 will receive the call <- no. by default that variable will be EMPTY and will generate an error when you try dialing it |
| 18:28.48 | qakhan | i did not show you full dialplan. but i used gotoif variable is empty then dial 8000 |
| 18:30.14 | [TK]D-Fender | Still bad that it'd forget |
| 18:31.28 | qakhan | [TK]D-Fender OK. i forget joe, i make you happy by doing AstDB ;) |
| 18:32.01 | qakhan | i respect you alot |
| 18:33.09 | qakhan | i will send you config after lunch |
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| 19:28.13 | qakhan | [TK]D-Fender here is my config with AstDB http://pastebin.com/GcnhBWB2 |
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| 19:52.32 | matt_ | does anybody know if e164.org should be working? |
| 19:53.33 | [TK]D-Fender | qakhan, clearly no good, and a wasted Set() as well |
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| 20:02.40 | xpheres | hei, anybody has succeded configuring SRTP encryption? |
| 20:04.16 | [TK]D-Fender | Asking the same question only vaguer won't get you anywhere further... |
| 20:05.31 | robmal | Who was this SRTP and why do you want to encrypt anything his way? |
| 20:05.56 | qakhan | [TK]D-Fender what is wrong now? |
| 20:07.15 | [TK]D-Fender | qakhan, Look at the values you are using |
| 20:07.17 | xpheres | TK Fender, I gave details today morning and I copied a log in pastebin |
| 20:07.18 | [TK]D-Fender | they do not MATCH |
| 20:07.42 | xpheres | I activate SRTP encryption in some extensions and I tried to log in and make calls from zoiper |
| 20:07.49 | [TK]D-Fender | xpheres, Who do you think is going to assume this, scroll back, find them, and hope that the posts haven't expired, etc? |
| 20:08.07 | xpheres | whatever I do, I'm not able to log in or make calls when I activate the SRTP in the extension and I don't know why |
| 20:08.12 | [TK]D-Fender | xpheres, Don't assume people are going to scroll back for hours of logs hoping to find something.\ |
| 20:08.16 | qakhan | [TK]D-Fender you are talking about this exten => 8000,1,Set(disp-ext=${DB(sas-disp/ext)}) |
| 20:08.22 | xpheres | ok I will copy it again |
| 20:08.52 | qakhan | i fixed it |
| 20:17.42 | xpheres | here is my log: http://pastebin.com/qjxsMYWZ |
| 20:28.30 | [TK]D-Fender | [2015-09-19 21:16:58] DEBUG[1379][C-00000045]: chan_sip.c:25730 handle_request_invite: No compatible codecs for this SIP call. |
| 20:28.44 | [TK]D-Fender | I see a VERY clear looking error about NO CODECS they can agree on there |
| 20:30.24 | [TK]D-Fender | [2015-09-19 21:16:58] WARNING[1379][C-00000045]: chan_sip.c:10446 process_sdp: Matched device setup to use SRTP, but request was not! |
| 20:30.47 | [TK]D-Fender | And th phone was not ASKING for SRTP. Asterisk IS apparently. |
| 20:30.52 | xpheres | The codecs works when I disable SRTP on extensions |
| 20:31.06 | [TK]D-Fender | So that says that you have not configured the phone to use it |
| 20:31.07 | xpheres | I configured zoiper to enable SRTP |
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| 20:31.18 | xpheres | on an android device and on linux |
| 20:31.59 | xpheres | but on the android device when I select TLS as transport I can not even log in |
| 20:32.35 | xpheres | I just set up again SRTP to no and everything works: codecs works, calls works everything |
| 20:32.48 | xpheres | I don't know what I am missing |
| 20:34.12 | [TK]D-Fender | TLS != SRTP |
| 20:34.19 | xpheres | I know |
| 20:34.28 | [TK]D-Fender | Don't mix the topics |
| 20:34.37 | [TK]D-Fender | it says that the sides are NOT asking for the same thing |
| 20:34.39 | [TK]D-Fender | it means it |
| 20:34.55 | [TK]D-Fender | [2015-09-19 21:16:58] WARNING[1379][C-00000045]: chan_sip.c:10446 process_sdp: Matched device setup to use SRTP, but request was not! <----------- |
| 20:34.56 | xpheres | but in zoiper I receives the following message everytime I enable SRTP: "TLS should be enable for SRTP, do you want to enable TLS now?" |
| 20:35.10 | [TK]D-Fender | Why should I CARE about that? |
| 20:35.20 | [TK]D-Fender | "should" does NOT mean "have to" |
| 20:35.43 | [TK]D-Fender | So if YOU go and start making other changes that are not agreed by both sides, then that is your responsibility |
| 20:35.43 | xpheres | so can I use srtp over UDP? |
| 20:35.51 | xpheres | ok... |
| 20:36.02 | [TK]D-Fender | RTPS and SRTP are always over UDP. |
| 20:36.06 | [TK]D-Fender | You missed the point\ |
| 20:36.11 | xpheres | I see |
| 20:36.31 | [TK]D-Fender | SIP is not REAL-TIME VOICE. It is just the signaling protocol and does natively and normally support both |
| 20:36.39 | xpheres | ok |
| 20:37.11 | xpheres | so using udp as transport I'm able to log in but I receives the message that the network is busy everytime I try to make a call |
| 20:38.05 | [TK]D-Fender | "network is busy" means nothing |
| 20:38.12 | [TK]D-Fender | that is not a real meaningful message |
| 20:38.16 | xpheres | ok |
| 20:38.43 | xpheres | I don't know what else can I do, I copied the log and I don't understand why I receive [2015-09-19 21:16:58] WARNING[1379][C-00000045]: chan_sip.c:10446 process_sdp: Matched device setup to use SRTP, but request was not! <----------- |
| 20:38.55 | [TK]D-Fender | it is very clear |
| 20:39.09 | [TK]D-Fender | the REQUESTER did not ask for SRTP |
| 20:39.11 | xpheres | when I enable it on both extensions and on the clients that connects to those extensions |
| 20:39.15 | [TK]D-Fender | What is unclear here? |
| 20:39.42 | xpheres | why did not ask for SRTP when it is enabled? |
| 20:40.00 | [TK]D-Fender | probably for being done wrong |
| 20:40.03 | xpheres | what should I do to make the device make the SRTP request? |
| 20:40.08 | xpheres | mmm |
| 20:40.20 | [TK]D-Fender | When it tells you this to your face you stop and go look. |
| 20:40.32 | [TK]D-Fender | OTHER settings missing? Certificates set up? |
| 20:40.34 | [TK]D-Fender | etc. |
| 20:40.39 | xpheres | mm |
| 20:40.47 | xpheres | maybe I should set up a certificate... |
| 20:41.08 | [TK]D-Fender | MAYBE? You need this for SRTP |
| 20:41.18 | xpheres | right |
| 20:41.32 | xpheres | sorry for the amateur ignorance, it is the first time I try to set up this |
| 20:44.06 | xpheres | ok, I'm using freepbx, I see in the menus there is a default certificate that was created when the software was installed |
| 20:44.25 | xpheres | so that means I have a certificate |
| 20:45.25 | xpheres | "A Certificate Authority is already present on this system. Deleting/Generating/Uploading will invalidate all of your current certificates!" |
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| 20:49.43 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 20:50.01 | [TK]D-Fender | Lets try this again.... |
| 20:50.29 | [TK]D-Fender | <[TK]D-Fender> the REQUESTER did not ask for SRTP |
| 20:50.32 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 20:55.54 | overyander | I'm having problems getting any softphone to register with a newly setup * box. i have my provider settings configured and am able to qualify with my provider. when looking at tcp dump of traffic of softphone register attempt, i see that each register request from the phone is immediately followed by a ICMP destination unreachable packet sent from my server to the outside of ip of the phone that just tried to register. |
| 20:57.27 | overyander | also, when i'm trying to register the phone, i don't see anything in * logs, not even sip debug |
| 21:02.58 | overyander | wow |
| 21:04.26 | overyander | so, the problem was that this being a public server (no natting) and having the bindaddr set to it's internal LAN IP (seperate NIC). i commented out the bindaddr and only left the externip defined and now it works. |
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| 21:07.12 | overyander | i just set the bindaddr, tcpbindaddr and tlsbindaddr to the external IP and after a sip reload it still works. i assume that the tcpbindaddr and tlsbindaddr will be required for those techs? |
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